20
Asterisk (FreePBX) Kein Raustelefonieren möglich "Busy"
Hallo alle zusammen,
und zwar harbe ich ein Problem. Ich habe einen VOIP-Telekom-Anschluss und möchte jetzt Asterisk als VOIP-Server nutzen.
Als Router wird Pfsense genutzt, Ports werden zu dem Asterisk-Server weitergeleitet.
Rein telefonieren kann ich.
Pfsense Nat Port-Forward:
Wenn ich jetzt raus telefonieren möchte, bekomme ich die Ansage das der Anruf "busy" ist.
Errorlog:
extensions.con:
extensions_additional.conf
pjsip.aor.conf:
pjsip.auth.conf
pjsip.conf
pjsip.endpoint.conf
pjsip.identify.conf
pjsip.registration.conf
pjsip.transports.conf
pjsip_notify.conf
Vor diesem Versuch hatte ich Asterisk über Debian 8 installiert und konnte da nur max 1min telefonieren, rein und raus!!!
Es muss irgendetwas an der Config sein, ich habe auch keine Ahnung was.
und zwar harbe ich ein Problem. Ich habe einen VOIP-Telekom-Anschluss und möchte jetzt Asterisk als VOIP-Server nutzen.
Als Router wird Pfsense genutzt, Ports werden zu dem Asterisk-Server weitergeleitet.
Rein telefonieren kann ich.
Pfsense Nat Port-Forward:
Wenn ich jetzt raus telefonieren möchte, bekomme ich die Ansage das der Anruf "busy" ist.
Errorlog:
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [+49{Telefonnummer Extern}@from-internal:1] Macro("PJSIP/300-00000012", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/300-00000012", "TOUCH_MONITOR=1477514006.92") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/300-00000012", "AMPUSER=300") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/300-00000012", "0?report") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("PJSIP/300-00000012", "1?Set(REALCALLERIDNUM=300)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:5] Set("PJSIP/300-00000012", "AMPUSER=300") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("PJSIP/300-00000012", "0?limit") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:7] Set("PJSIP/300-00000012", "AMPUSERCIDNAME=User") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("PJSIP/300-00000012", "0?report") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:9] Set("PJSIP/300-00000012", "AMPUSERCID=300") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:10] Set("PJSIP/300-00000012", "__DIAL_OPTIONS=Ttr") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:11] Set("PJSIP/300-00000012", "CALLERID(all)="User" <300>") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:12] GotoIf("PJSIP/300-00000012", "0?limit") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:13] ExecIf("PJSIP/300-00000012", "1?Set(GROUP(concurrency_limit)=300)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:14] GosubIf("PJSIP/300-00000012", "7?sub-ccss,s,1(from-internal,)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-ccss:1] ExecIf("PJSIP/300-00000012", "0?Return()") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-ccss:2] Set("PJSIP/300-00000012", "CCSS_SETUP=TRUE") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-ccss:3] GosubIf("PJSIP/300-00000012", "0?monitor_config,1(from-internal,):monitor_default,1(from-internal,)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [monitor_default@sub-ccss:1] GotoIf("PJSIP/300-00000012", "0?is_exten") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [monitor_default@sub-ccss:2] StackPop("PJSIP/300-00000012", "") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [monitor_default@sub-ccss:3] Return("PJSIP/300-00000012", "FALSE") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:15] ExecIf("PJSIP/300-00000012", "0?Set(CHANNEL(language)=)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:16] GotoIf("PJSIP/300-00000012", "1?continue") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (macro-user-callerid,s,30)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:30] Set("PJSIP/300-00000012", "CALLERID(number)=300") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:31] Set("PJSIP/300-00000012", "CALLERID(name)=User") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:32] Set("PJSIP/300-00000012", "CDR(cnum)=300") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:33] Set("PJSIP/300-00000012", "CDR(cnam)=User") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-user-callerid:34] Set("PJSIP/300-00000012", "CHANNEL(language)=en") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [+49{Telefonnummer Extern}@from-internal:2] Gosub("PJSIP/300-00000012", "sub-record-check,s,1(out,+49{Telefonnummer Extern},dontcare)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:1] GotoIf("PJSIP/300-00000012", "0?initialized") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:2] Set("PJSIP/300-00000012", "__REC_STATUS=INITIALIZED") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:3] Set("PJSIP/300-00000012", "NOW=1477514006") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:4] Set("PJSIP/300-00000012", "__DAY=26") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:5] Set("PJSIP/300-00000012", "__MONTH=10") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:6] Set("PJSIP/300-00000012", "__YEAR=2016") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:7] Set("PJSIP/300-00000012", "__TIMESTR=20161026-223326") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:8] Set("PJSIP/300-00000012", "__FROMEXTEN=300") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:9] Set("PJSIP/300-00000012", "__MON_FMT=wav") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:10] NoOp("PJSIP/300-00000012", "Recordings initialized") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:11] ExecIf("PJSIP/300-00000012", "0?Set(ARG3=dontcare)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:12] Set("PJSIP/300-00000012", "REC_POLICY_MODE_SAVE=") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:13] ExecIf("PJSIP/300-00000012", "0?Set(REC_STATUS=NO)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:14] GotoIf("PJSIP/300-00000012", "3?checkaction") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (sub-record-check,s,17)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@sub-record-check:17] GotoIf("PJSIP/300-00000012", "1?sub-record-check,out,1") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (sub-record-check,out,1)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [out@sub-record-check:1] NoOp("PJSIP/300-00000012", "Outbound Recording Check from 300 to +49{Telefonnummer Extern}") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [out@sub-record-check:2] Set("PJSIP/300-00000012", "RECMODE=dontcare") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [out@sub-record-check:3] ExecIf("PJSIP/300-00000012", "1?Goto(routewins)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (sub-record-check,out,7)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [out@sub-record-check:7] Gosub("PJSIP/300-00000012", "recordcheck,1(dontcare,out,+49{Telefonnummer Extern})") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/300-00000012", "Starting recording check against dontcare") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("PJSIP/300-00000012", "dontcare") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [recordcheck@sub-record-check:3] Return("PJSIP/300-00000012", "") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [out@sub-record-check:8] Return("PJSIP/300-00000012", "") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [+49{Telefonnummer Extern}@from-internal:3] ExecIf("PJSIP/300-00000012", "0 ?Set(CDR(accountcode)=)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [+49{Telefonnummer Extern}@from-internal:4] Set("PJSIP/300-00000012", "MOHCLASS=default") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [+49{Telefonnummer Extern}@from-internal:5] ExecIf("PJSIP/300-00000012", "1?Set(TRUNKCIDOVERRIDE="User" <0{Telefonnummer Intern}>)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [+49{Telefonnummer Extern}@from-internal:6] Set("PJSIP/300-00000012", "_NODEST=") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [+49{Telefonnummer Extern}@from-internal:7] Macro("PJSIP/300-00000012", "dialout-trunk,2,+49{Telefonnummer Extern},,off") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:1] Set("PJSIP/300-00000012", "DIAL_TRUNK=2") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:2] GosubIf("PJSIP/300-00000012", "0?sub-pincheck,s,1()") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:3] GotoIf("PJSIP/300-00000012", "0?disabletrunk,1") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:4] Set("PJSIP/300-00000012", "DIAL_NUMBER=+49{Telefonnummer Extern}") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:5] Set("PJSIP/300-00000012", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:6] Set("PJSIP/300-00000012", "OUTBOUND_GROUP=OUT_2") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:7] GotoIf("PJSIP/300-00000012", "0?nomax") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:8] GotoIf("PJSIP/300-00000012", "0?chanfull") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:9] GotoIf("PJSIP/300-00000012", "0?skipoutcid") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:10] Set("PJSIP/300-00000012", "DIAL_TRUNK_OPTIONS=Tt") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:11] Macro("PJSIP/300-00000012", "outbound-callerid,2") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:1] ExecIf("PJSIP/300-00000012", "0?Set(CALLERPRES()=)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:2] ExecIf("PJSIP/300-00000012", "0?Set(REALCALLERIDNUM=300)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:3] GotoIf("PJSIP/300-00000012", "1?normcid") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (macro-outbound-callerid,s,6)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:6] Set("PJSIP/300-00000012", "USEROUTCID=") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:7] Set("PJSIP/300-00000012", "EMERGENCYCID=") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:8] Set("PJSIP/300-00000012", "TRUNKOUTCID=0{Telefonnummer Intern}") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:9] GotoIf("PJSIP/300-00000012", "1?trunkcid") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (macro-outbound-callerid,s,14)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:14] ExecIf("PJSIP/300-00000012", "1?Set(CALLERID(all)=0{Telefonnummer Intern})") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:15] ExecIf("PJSIP/300-00000012", "0?Set(CALLERID(all)=)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:16] ExecIf("PJSIP/300-00000012", "1?Set(CALLERID(all)=User <0{Telefonnummer Intern}>)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:17] ExecIf("PJSIP/300-00000012", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:18] Set("PJSIP/300-00000012", "CDR(outbound_cnum)=0{Telefonnummer Intern}") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:19] Set("PJSIP/300-00000012", "CDR(outbound_cnam)=User") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:12] GosubIf("PJSIP/300-00000012", "0?sub-flp-2,s,1()") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:13] Set("PJSIP/300-00000012", "OUTNUM=+49{Telefonnummer Extern}") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:14] Set("PJSIP/300-00000012", "custom=PJSIP") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:15] ExecIf("PJSIP/300-00000012", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:16] ExecIf("PJSIP/300-00000012", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:17] Macro("PJSIP/300-00000012", "dialout-trunk-predial-hook,") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/300-00000012", "") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/300-00000012", "0?bypass,1") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/300-00000012", "1?Set(CONNECTEDLINE(num,i)=+49{Telefonnummer Extern})") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/300-00000012", "1?Set(CONNECTEDLINE(name,i)=CID:0{Telefonnummer Intern})") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:21] GotoIf("PJSIP/300-00000012", "0?customtrunk") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:22] Dial("PJSIP/300-00000012", "PJSIP/+49{Telefonnummer Extern}@Telekom,300,Tt") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] app_dial.c: Called PJSIP/+49{Telefonnummer Extern}@Telekom
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:23] NoOp("PJSIP/300-00000012", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:24] GotoIf("PJSIP/300-00000012", "0?continue,1:s-CHANUNAVAIL,1") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("PJSIP/300-00000012", "RC=21") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("PJSIP/300-00000012", "21,1") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (macro-dialout-trunk,21,1)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [21@macro-dialout-trunk:1] Goto("PJSIP/300-00000012", "continue,1") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx_builtins.c: Goto (macro-dialout-trunk,continue,1)
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [continue@macro-dialout-trunk:1] NoOp("PJSIP/300-00000012", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [continue@macro-dialout-trunk:2] Set("PJSIP/300-00000012", "CALLERID(number)=300") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [+49{Telefonnummer Extern}@from-internal:8] Macro("PJSIP/300-00000012", "outisbusy,") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outisbusy:1] Progress("PJSIP/300-00000012", "") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outisbusy:2] GotoIf("PJSIP/300-00000012", "0?emergency,1") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outisbusy:3] GotoIf("PJSIP/300-00000012", "0?intracompany,1") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outisbusy:4] Playback("PJSIP/300-00000012", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[2016-10-26 22:33:26] VERBOSE[22441][C-00000009] file.c: <PJSIP/300-00000012> Playing 'all-circuits-busy-now.ulaw' (language 'en')
[2016-10-26 22:33:28] VERBOSE[22441][C-00000009] file.c: <PJSIP/300-00000012> Playing 'pls-try-call-later.ulaw' (language 'en')
[2016-10-26 22:33:30] VERBOSE[22441][C-00000009] pbx.c: Executing [s@macro-outisbusy:5] Congestion("PJSIP/300-00000012", "20") in new stack
[2016-10-26 22:33:30] WARNING[22441][C-00000009] channel.c: Prodding channel 'PJSIP/300-00000012' failed
[2016-10-26 22:33:30] VERBOSE[22441][C-00000009] app_macro.c: Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'PJSIP/300-00000012' in macro 'outisbusy'
[2016-10-26 22:33:30] VERBOSE[22441][C-00000009] pbx.c: Spawn extension (from-internal, +49{Telefonnummer Extern}, 8) exited non-zero on 'PJSIP/300-00000012'
[2016-10-26 22:33:30] VERBOSE[22441][C-00000009] pbx.c: Executing [h@from-internal:1] Hangup("PJSIP/300-00000012", "") in new stack
[2016-10-26 22:33:30] VERBOSE[22441][C-00000009] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/300-00000012'
[2016-10-26 22:33:32] WARNING[13227] res_pjsip_pubsub.c: No registered publish handler for event presence
[2016-10-26 22:33:32] WARNING[13227] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfoextensions.con:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;*******************************************************************************
; AUTO-GENERATED AND CUSOTM USER DIALPLAN INCLUDED HERE *
;*******************************************************************************
;--------------------------------------------------------------------------------;
; Customizations to this dialplan should be made in extensions_custom.conf
; See extensions_custom.conf.sample for an example.
;
; If you need to use [macro-dialout-trunk-predial-hook], [ext-did-custom], or
; [from-internal-custom] for example, place these in this file or they will get overwritten.
;
; WARNING ABOUT: #include extensions_override_freepbx.conf
;
; This include file is put first to allow the auto-generated dialplan in FreePBX
; to be overwritten if necessary. Overriding auto-generated dialplan should be done
; with extreme caution. In almost all cases any custom dialplan SHOULD be put in
; extensions_custom.conf which will not hurt a FreePBX generated dialplan. In some
; very rare and custom situations users may have a need to override what FreePBX
; automatically generates. If so anything in this file will do that. If you come up
; with situations where you need to modify the existing dialplan or macros, put it
; here.
;
#include extensions_override_freepbx.conf
#include extensions_additional.conf
#include extensions_custom.conf
;--------------------------------------------------------------------------------;
;*******************************************************************************
; DIALPLAN OPEN TO THE PUBLIC, USING INBOUND DIDS AND SIMILAR *
;*******************************************************************************
;
;-------------------------------------------------------------------------------
; from-digital:
;
; Context to set for PRI's and equivalent
;
[from-digital]
include => from-pstn
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; from-analog:
;
; Context to set for analog DAHDi cards and equivalent
;
[from-analog]
include => from-dahdi
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; from-trunk:
;
; Context is really just an aliax of from-pstn
;
[from-trunk]
include => from-pstn
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; from-pstn:
;
; Entry context for calls from the outside world to hit FreePBX
[from-pstn]
include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations
include => ext-did
include => ext-did-post-custom
include => from-did-direct
include => ext-did-catchall ; THIS MUST COME AFTER ext-did
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; from-pstn-e164-us:
;
; The context is designed for providers who send calls in e164 format and is
; biased towards NPA calls, callerid and dialing rules. It will do the following:
;
; DIDs in an NPA e164 format of +1NXXNXXXXXX will be converted to 10 digit DIDs
;
; DIDs in any other format will be delivered as they are, including e164 non NPA
; DIDs which means they will need the full format including the + in the inbound
; route.
;
; CallerID(number) presented in e164 NPA format will be trimmed to a 10 digit CID
;
; CallerID(number) presented in e164 non-NPA (country code other than 1) will be
; reformated from: +<CountryCode><Number> to 011<CountryCode><Number>
;
[from-pstn-e164-us]
exten => _+1NXXNXXXXXX/_+1NXXNXXXXXX,1,Set(CALLERID(number)=${CALLERID(number):2})
exten => _+1NXXNXXXXXX/_NXXNXXXXXX,2,Goto(from-pstn,${EXTEN:2},1)
exten => _+1NXXNXXXXXX/_+NX.,1,Set(CALLERID(number)=011${CALLERID(number):1})
exten => _+1NXXNXXXXXX/_011NX.,n,Goto(from-pstn,${EXTEN:2},1)
exten => _+1NXXNXXXXXX,1,Goto(from-pstn,${EXTEN:2},1)
exten => _[0-9+]./_+1NXXNXXXXXX,1,Set(CALLERID(number)=${CALLERID(number):2})
exten => _[0-9+]./_1NXXNXXXXXX,1,Set(CALLERID(number)=${CALLERID(number):1})
exten => _[0-9+]./_NXXNXXXXXX,n,Goto(from-pstn,${EXTEN},1)
exten => _[0-9+]./_+NX.,1,Set(CALLERID(number)=011${CALLERID(number):1})
exten => _[0-9+]./_011NX.,n,Goto(from-pstn,${EXTEN},1)
exten => _[0-9+].,1,Goto(from-pstn,${EXTEN},1)
exten => s/_+1NXXNXXXXXX,1,Set(CALLERID(number)=${CALLERID(number):2})
exten => s/_NXXNXXXXXX,n,Goto(from-pstn,${EXTEN},1)
exten => s/_+NX.,1,Set(CALLERID(number)=011${CALLERID(number):1})
exten => s/_011NX.,n,Goto(from-pstn,${EXTEN},1)
exten => s,1,Goto(from-pstn,${EXTEN},1)
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; from-pstn-to-did:
;
; The context is designed for providers who send the DID in the TO: SIP header
; only. The format of this header is:
;
; To: <sip:2125551212@172.31.74.25>
;
; So the DID must be extracted between the sip: and the @, which this does
;
[from-pstn-toheader]
exten => _.,1,Goto(from-pstn,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; from-sip-external
;
; This context is the default SIP context unless otherwise changed in the SIP
; Settings module or other sip configuration locations. This context is hit by
; either anonymous SIP calls or mis-configured SIP trunks when the incoming call
; can not be matched with a SIP section.
;
[from-sip-external]
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?checklang:noanonymous)
exten => s,n(checklang),GotoIf($["${SIPLANG}"!=""]?setlanguage:from-trunk,${DID},1)
exten => s,n(setlanguage),Set(CHANNEL(language)=${SIPLANG})
exten => s,n,Goto(from-trunk,${DID},1)
exten => s,n(noanonymous),Set(TIMEOUT(absolute)=15)
exten => s,n,Log(WARNING,"Rejecting unknown SIP connection from ${CHANNEL(recvip)}")
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
;-------------------------------------------------------------------------------
;*******************************************************************************
; INTERNAL DIALPLAN, NOT OPEN TO THE PUBLIC WORLD *
;*******************************************************************************
;
;-------------------------------------------------------------------------------
; from-internal:
;
; Internal dialplan that most internal phones have access to
;
[from-internal]
include => from-internal-noxfer
include => from-internal-xfer
include => bad-number ; auto-generated
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; from-internal-noxfer:
;
; Place to put internal dialplan that should not be accessible during a blind
; transfer, this context will not be visible during such.
;
[from-internal-noxfer]
include => from-internal-noxfer-custom
include => from-internal-noxfer-additional ; auto-generated
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; from-internal-xfer:
;
; Place to put most internal dialplan, will be visible during normal calls and
; blind transfers.
;
[from-internal-xfer]
include => from-internal-custom
include => from-internal-additional ; auto-generated
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)
;-------------------------------------------------------------------------------
;*******************************************************************************
; INTERNAL MACROS & SPECIAL CONTEXTS *
;*******************************************************************************
;
;-------------------------------------------------------------------------------
; from-did-direct:
;
; forces ext-findmefollow to take precedence over ext-local. Also exposed to
; the public side to allow an extension number to be used as an external DID
; without requiring inbound routes to be created, common in many PRI installations
; where the last 4 digits are used as the extnension and DIDs are delivered in
; 4 digit formats.
;
[from-did-direct]
include => ext-findmefollow
include => ext-local
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-dial:
;
; Rings one or more extensions. Handles things like call forwarding and DND
; We don't call dial directly for anything internal anymore.
; ARGS: $TIMER, $OPTIONS, $EXT1, $EXT2, $EXT3, ...
; Use a Macro call such as the following:
;
; Macro(dial,$DIAL_TIMER,$DIAL_OPTIONS,$EXT1,$EXT2,$EXT3,...)
;
[macro-dial]
exten => s,1,GotoIf($["${MOHCLASS}" = ""]?dial)
exten => s,n,Set(CHANNEL(musicclass)=${MOHCLASS})
exten => s,n(dial),AGI(dialparties.agi)
exten => s,n,NoOp(Returned from dialparties with no extensions to call and DIALSTATUS: ${DIALSTATUS})
exten => s,n+2(normdial),Dial(${ds}) ; dialparties will set the priority to 10 if $ds is not null
exten => s,n,Set(DIALSTATUS=${IF($["${DIALSTATUS_CW}"!="" ]?${DIALSTATUS_CW}:${DIALSTATUS})})
exten => s,n,GosubIf($[("${SCREEN}" != "" & ("${DIALSTATUS}" = "TORTURE" | "${DIALSTATUS}" = "DONTCALL")) | "${DIALSTATUS}" = "ANSWER"]?${DIALSTATUS},1)
exten => s,20(huntdial),NoOp(Returned from dialparties with hunt groups to dial )
exten => s,n,Set(HuntLoop=0)
exten => s,n(a22),GotoIf($[${HuntMembers} >= 1]?a30) ; if this is from rg-group, don't strip prefix
exten => s,n,NoOp(Returning there are no members left in the hunt group to ring)
; dialparties.agi has setup the dialstring for each hunt member in a variable labeled HuntMember0, HuntMember1 etc for each iteration
; and The total number in HuntMembers. So for each iteration, we will update the CALLTRACE Data.
;
exten => s,n+2(a30),Set(HuntMember=HuntMember${HuntLoop})
exten => s,n,GotoIf($[$["${CALLTRACE_HUNT}" != "" ] & $[$["${RingGroupMethod}" = "hunt" ] | $["${RingGroupMethod}" = "firstavailable"] | $["${RingGroupMethod}" = "firstnotonphone"]]]?a32:a35)
exten => s,n(a32),Set(CT_EXTEN=${CUT(FILTERED_DIAL,,$[${HuntLoop} + 1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Goto(s,a42)
;Set Call Trace for each hunt member we are going to call "Memory groups have multiple members to set CALL TRACE For" hence the loop
;
exten => s,n(a35),GotoIf($[$["${CALLTRACE_HUNT}" != "" ] & $["${RingGroupMethod}" = "memoryhunt" ]]?a36:a50)
exten => s,n(a36),Set(CTLoop=0)
exten => s,n(a37),GotoIf($[${CTLoop} > ${HuntLoop}]?a42) ; if this is from rg-group, don't strip prefix
exten => s,n,Set(CT_EXTEN=${CUT(FILTERED_DIAL,,$[${CTLoop} + 1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Set(CTLoop=$[1 + ${CTLoop}])
exten => s,n,Goto(s,a37)
exten => s,n(a42),Dial(${${HuntMember}}${ds})
exten => s,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?ANSWER,1)
exten => s,n,Set(HuntLoop=$[1 + ${HuntLoop}])
exten => s,n,GotoIf($[$[$["foo${RingGroupMethod}" != "foofirstavailable"] & $["foo${RingGroupMethod}" != "foofirstnotonphone"]] | $["foo${DialStatus}" = "fooBUSY"]]?a46)
exten => s,n,Set(HuntMembers=0)
exten => s,n(a46),Set(HuntMembers=$[${HuntMembers} - 1])
exten => s,n,Goto(s,a22)
exten => s,n(a50),Noop(Deleting: CALLTRACE/${CT_EXTEN} ${DB_DELETE(CALLTRACE/${CT_EXTEN})})
exten => s,n,Goto(s,a42)
; For call screening
exten => NOANSWER,1,Macro(vm,${SCREEN_EXTEN},BUSY,${IVR_RETVM})
exten => NOANSWER,n,GotoIf($["${IVR_RETVM}" != "RETURN" | "${IVR_CONTEXT}" = ""]?bye)
exten => NOANSWER,n,Return
exten => NOANSWER,n(bye),Macro(hangupcall)
exten => TORTURE,1,Goto(app-blackhole,musiconhold,1)
exten => TORTURE,n,Macro(hangupcall)
exten => DONTCALL,1,Answer
exten => DONTCALL,n,Wait(1)
exten => DONTCALL,n,Zapateller()
exten => DONTCALL,n,Playback(ss-noservice)
exten => DONTCALL,n,Macro(hangupcall)
exten => ANSWER,1,Noop(Call successfully answered - Hanging up now)
exten => ANSWER,n,Macro(hangupcall,)
; make sure hungup calls go here so that proper cleanup occurs from call confirmed calls and the like
;
exten => h,1,Macro(hangupcall)
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-get-vmcontext:
;
; returns the user's voicemail context
;
[macro-get-vmcontext]
exten => s,1,Set(VMCONTEXT=${DB(AMPUSER/${ARG1}/voicemail)})
exten => s,2,GotoIf($["foo${VMCONTEXT}" = "foo"]?200:300)
exten => s,200,Set(VMCONTEXT=default)
exten => s,300,NoOp()
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-outisbusy:
;
; called if all trunks are busy. This macro is overwridden by the
; Outbound Route Messages module if installed and configured to provide
; better messages or choices of tones.
;
[macro-outisbusy]
exten => s,1,Progress
exten => s,n,Playback(all-circuits-busy-now,noanswer)
exten => s,n,Playback(pls-try-call-later,noanswer)
exten => s,n,Macro(hangupcall)
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-dialout-trunk-predial-hook:
;
; this macro intentionally left blank so it may be safely overwritten for any custom
; requirements that an installation may have.
;
; the macro is called by macro-dialout-trunk just prior to making a Dial() attempt
; to a trunk.
;
; MACRO RETURN CODE: ${PREDIAL_HOOK_RET}
; if set to "BYPASS" then this trunk will be skipped
;
;
[macro-dialout-trunk-predial-hook]
exten => s,1,MacroExit()
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-dialout-one-predial-hook
;
; this macro intentionally left blank so it may be safely overwritten for any custom
; requirements that an installation may have.
;
; the macro is called by macro-dialout-trunk just prior to making a Dial() attempt
; to an individual extension.
;
[macro-dialout-one-predial-hook]
exten => s,1,MacroExit()
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-dialout-dundi-predial-hook:
;
; this macro intentionally left blank so it may be safely overwritten for any custom
; requirements that an installation may have.
;
; the macro is called by macro-dialout-dundi just prior to making a Dial() attempt
; to a trunk.
;
; MACRO RETURN CODE: ${PREDIAL_HOOK_RET}
; if set to "BYPASS" then this trunk will be skipped
;
;
[macro-dialout-dundi-predial-hook]
exten => s,1,MacroExit()
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-user-logon:
;
; Used to log a user onto an adhoc device. Most of the work is done by
; user_login_out.agi AGI script
;
[macro-user-logon]
exten => s,1,Set(DEVICETYPE=${DB(DEVICE/${CALLERID(number)}/type)})
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,GotoIf($["${DEVICETYPE}" = "fixed"]?s-FIXED,1)
; get user's extension
;
exten => s,n,Set(AMPUSER=${ARG1})
exten => s,n,GotoIf($["${AMPUSER}" != ""]?gotpass)
exten => s,n(playagain),Read(AMPUSER,please-enter-your-extension-then-press-pound,,,4)
; get user's password and authenticate
;
exten => s,n,GotoIf($["${AMPUSER}" = ""]?s-MAXATTEMPTS,1)
exten => s,n(gotpass),GotoIf($["${DB_EXISTS(AMPUSER/${AMPUSER}/password)}" = "0"]?s-NOUSER,1)
exten => s,n,Set(AMPUSERPASS=${DB_RESULT})
exten => s,n,GotoIf($[${LEN(${AMPUSERPASS})} = 0]?s-NOPASSWORD,1)
; do not continue if the user has already logged onto this device
;
exten => s,n,Set(DEVICEUSER=${DB(DEVICE/${CALLERID(number)}/user)})
exten => s,n,GotoIf($["${DEVICEUSER}" = "${AMPUSER}"]?s-ALREADYLOGGEDON,1)
exten => s,n,Authenticate(${AMPUSERPASS})
exten => s,n,AGI(user_login_out.agi,login,${CALLERID(number)},${AMPUSER})
exten => s,n,Playback(agent-loginok)
exten => s-FIXED,1,NoOp(Device is FIXED and cannot be logged into)
exten => s-FIXED,n,Playback(ha/phone)
exten => s-FIXED,n,SayDigits(${CALLERID(number)})
exten => s-FIXED,n,Playback(is-curntly-unavail&vm-goodbye)
exten => s-FIXED,n,Hangup ;TODO should play msg indicated device cannot be logged into
exten => s-ALREADYLOGGEDON,1,NoOp(This device has already been logged into by this user)
exten => s-ALREADYLOGGEDON,n,Playback(vm-goodbye)
exten => s-ALREADYLOGGEDON,n,Hangup ;TODO should play msg indicated device is already logged into
exten => s-NOPASSWORD,1,NoOp(This extension does not exist or no password is set)
exten => s-NOPASSWORD,n,Playback(pbx-invalid)
exten => s-NOPASSWORD,n,Goto(s,playagain)
exten => s-MAXATTEMPTS,1,NoOp(Too many login attempts)
exten => s-MAXATTEMPTS,n,Playback(vm-goodbye)
exten => s-MAXATTEMPTS,n,Hangup
exten => s-NOUSER,1,NoOp(Invalid extension ${AMPUSER} entered)
exten => s-NOUSER,n,Playback(pbx-invalid)
exten => s-NOUSER,n,Goto(s,playagain)
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-user-logoff:
;
; Used to log a user off of an adhoc device. Most of the work is done by
; user_login_out.agi AGI script
;
[macro-user-logoff]
exten => s,1,Set(DEVICETYPE=${DB(DEVICE/${CALLERID(number)}/type)})
exten => s,n,GotoIf($["${DEVICETYPE}" = "fixed"]?s-FIXED,1)
exten => s,n,AGI(user_login_out.agi,logout,${CALLERID(number)})
exten => s,n(done),Playback(agent-loggedoff)
exten => s-FIXED,1,NoOp(Device is FIXED and cannot be logged out of)
exten => s-FIXED,n,Playback(an-error-has-occured&vm-goodbye)
exten => s-FIXED,n,Hangup ;TODO should play msg indicated device cannot be logged into
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; default
;
; FreePBX does not use the default context. This context is used by asterisk when
; it has no other information provided and needs to deliver a call. Hitting this means
; there has been some sort of configuration error, or a potential bug somehwere.
; This context can be reached from either internal or external sources.
;
[default]
include => ext-local
exten => s,1,Playback(vm-goodbye)
exten => s,n,Noop(ERROR: FreePBX Does not use the [default] context, confguration error)
exten => s,n,Macro(hangupcall)
;-------------------------------------------------------------------------------
;*******************************************************************************
; DEPRECATED DIALPLAN - THESE WILL BE REMOVED, NOT USED BY FREEPBX *
;*******************************************************************************
;
;-------------------------------------------------------------------------------
; macro-fixcid: [DEPRECATED]
;
; For some reason, if I don't run setCIDname, CALLERID(name) will be blank in my AGI
;
[macro-fixcid]
exten => s,1,Set(CALLERID(name)=${CALLERID(name)})
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-sayXXXX: [DEPRECATED]
;
; Text-To-Speech related macros
; These all follow common actions. First try to playback a file "tts/custom-md5"
; where "md5" is the md5() of whatever is going to be played. If that doesn't exist,
; try to playback using macro-tts-sayXXXXX (where XXXXX is text/digits/etc, same as
; the macro below). If that macro exits with MACRO_OFFSET=100, then it's done,
; otherwise, fallback to the default asterisk method.
;
;-------------------------------------------------------------------------------
; macro-saytext: [DEPRECATED]
;
; say text is purely for text-to-speech, there is no fallback
;
[macro-saytext]
exten => s,1,Noop(Trying custom SayText playback for "${ARG1}")
exten => s,n,Playback(tts/custom-${MD5(${ARG1})})
exten => s,n,GotoIf($["${PLAYBACKSTATUS}"="SUCCESS"]?done)
; call tts-saytext. This should set MACRO_OFFSET=101 if it was successful
exten => s,n(tts),Macro(tts-saytext,${ARG1},${ARG2},${ARG3})
exten => s,n,Noop(No text-to-speech handler for SayText, cannot say "${ARG1}")
exten => s,n,Goto(done)
exten => s,tts+101,Noop(tts handled saytext)
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-sayname: [DEPRECATED]
;
; say name is for saying names typically, but fallsback to using SayAlpha
; (saying the word letter-by-letter)
;
[macro-sayname]
exten => s,1,Noop(Trying custom SayName playback for "${ARG1}")
exten => s,n,Playback(tts/custom-${MD5(${ARG1})})
exten => s,n,GotoIf($["${PLAYBACKSTATUS}"="SUCCESS"]?done)
; call tts-sayalpha. This should set MACRO_OFFSET=101 if it was successful
exten => s,n(tts),Macro(tts-sayalpha,${ARG1},${ARG2},${ARG3})
exten => s,n,SayAlpha(${ARG1})
exten => s,n,Goto(done)
exten => s,tts+101,Noop(tts handled sayname)
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-saynumber: [DEPRECATED]
;
; Say number is for saying numbers (eg "one thousand forty six")
;
[macro-saynumber]
exten => s,1,Noop(Trying custom SayNumber playback for "${ARG1}")
exten => s,n,Playback(tts/custom-${MD5(${ARG1})})
exten => s,n,GotoIf($["${PLAYBACKSTATUS}"="SUCCESS"]?done)
; call tts-saynumber. This should set MACRO_OFFSET=101 if it was successful
exten => s,n(tts),Macro(tts-saynumber,${ARG1},${ARG2},${ARG3})
exten => s,n,SayNumber(${ARG1})
exten => s,n,Goto(done)
exten => s,tts+101,Noop(tts handled saynumber)
;-------------------------------------------------------------------------------
;-------------------------------------------------------------------------------
; macro-saydigits: [DEPRECATED]
;
; Say digits is for saying digits one-by-one (eg, "one zero four six")
;
[macro-saydigits]
exten => s,1,Noop(Trying custom SayDigits playback for "${ARG1}")
exten => s,n,Playback(tts/custom-${MD5(${ARG1})})
exten => s,n,GotoIf($["${PLAYBACKSTATUS}"="SUCCESS"]?done)
; call tts-saydigits. This should set MACRO_OFFSET=101 if it was successful
exten => s,n(tts),Macro(tts-saydigits,${ARG1},${ARG2},${ARG3})
exten => s,n,SayDigits(${ARG1})
exten => s,n,Goto(done)
;-------------------------------------------------------------------------------extensions_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
[globals]
CFDEVSTATE = TRUE
CAMPONTOGGLE = *84
DNDDEVSTATE = TRUE
FMDEVSTATE = TRUE
QUEDEVSTATE = TRUE
DYNAMIC_FEATURES = apprecord
INTERCOMCODE = *80
ASTETCDIR = /etc/asterisk
ASTMODDIR = /usr/lib/asterisk/modules
ASTVARLIBDIR = /var/lib/asterisk
ASTAGIDIR = /var/lib/asterisk/agi-bin
ASTSPOOLDIR = /var/spool/asterisk
ASTRUNDIR = /var/run/asterisk
ASTLOGDIR = /var/log/asterisk
CWINUSEBUSY = true
AMPMGRUSER = admin
AMPMGRPASS = amp111
AMPDBENGINE = mysql
AMPDBHOST = localhost
AMPDBNAME = asterisk
AMPDBUSER = freepbxuser
AMPDBPASS = 4afed5a03531
VMX_CONTEXT = from-internal
VMX_PRI = 1
VMX_TIMEDEST_CONTEXT =
VMX_TIMEDEST_EXT = dovm
VMX_TIMEDEST_PRI = 1
VMX_LOOPDEST_CONTEXT =
VMX_LOOPDEST_EXT = dovm
VMX_LOOPDEST_PRI = 1
MIXMON_DIR =
MIXMON_POST =
DIAL_OPTIONS = Ttr
TRUNK_OPTIONS = Tt
TRUNK_RING_TIMER = 300
MIXMON_FORMAT = wav
REC_POLICY = caller
RINGTIMER_DEFAULT = 15
TRANSFER_CONTEXT = from-internal-xfer
ASTVERSION = 13.9.1
ASTCHANDAHDI = 1
NULL = ""
OUT_2 = PJSIP
OUT_2_SUFFIX = @Telekom
OUTCID_2 = 04074041501
OUTMAXCHANS_2 = 2
OUTFAIL_2 =
OUTPREFIX_2 =
OUTDISABLE_2 = off
OUTKEEPCID_2 = off
FORCEDOUTCID_2 =
PREFIX_TRUNK_2 =
ALLOW_SIP_ANON = no
SIPLANG =
#include globals_custom.conf
;end of [globals]
[app-blacklist]
exten => s,1,Macro(user-callerid,)
exten => *32,1,Goto(app-blacklist-last,s,1)
exten => *31,1,Goto(app-blacklist-remove,s,1)
exten => *30,1,Goto(app-blacklist-add,s,1)
;--== end of [app-blacklist] ==--;
[app-blacklist-check]
exten => s,1(check),GotoIf($["${BLACKLIST()}"="1"]?blacklisted)
exten => s,n,Set(CALLED_BLACKLIST=1)
exten => s,n,Return()
exten => s,n(blacklisted),Answer
exten => s,n,Wait(1)
exten => s,n,Zapateller()
exten => s,n,Playback(ss-noservice)
exten => s,n,Hangup
;--== end of [app-blacklist-check] ==--;
[app-blacklist-last]
exten => s,1,Answer
exten => s,n,Macro(user-callerid,)
exten => s,n,Wait(1)
exten => s,n,Set(lastcaller=${DB(CALLTRACE/${AMPUSER})})
exten => s,n,GotoIf($[ $[ "${lastcaller}" = "" ] | $[ "${lastcaller}" = "unknown" ] ]?noinfo)
exten => s,n,Playback(privacy-to-blacklist-last-caller&telephone-number)
exten => s,n,SayDigits(${lastcaller})
exten => s,n,Set(TIMEOUT(digit)=3)
exten => s,n,Set(TIMEOUT(response)=7)
exten => s,n,Playback(if-correct-press&digits/1)
exten => s,n,Goto(end)
exten => s,n(noinfo),Playback(unidentified-no-callback)
exten => s,n,Hangup
exten => s,n,Noop(Waiting for input)
exten => s,n(end),WaitExten(60,)
exten => s,n,Playback(sorry-youre-having-problems&goodbye)
exten => 1,1,Set(DB(blacklist/${lastcaller})=1)
exten => 1,n,Playback(num-was-successfully)
exten => 1,n,Playback(added)
exten => 1,n,Wait(1)
exten => 1,n,Hangup
;--== end of [app-blacklist-last] ==--;
[app-blacklist-remove]
exten => s,1,Answer
exten => s,n,Macro(user-callerid,)
exten => s,n,Wait(1)
exten => s,n,Playback(entr-num-rmv-blklist)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=60)
exten => s,n,Read(blacknr,then-press-pound,,,,)
exten => s,n,SayDigits(${blacknr})
exten => s,n,Playback(if-correct-press&digits/1)
exten => s,n,Noop(Waiting for input)
exten => s,n(end),WaitExten(60,)
exten => s,n,Playback(sorry-youre-having-problems&goodbye)
exten => 1,1,Noop(Deleting: blacklist/${blacknr} ${DB_DELETE(blacklist/${blacknr})})
exten => 1,n,Playback(num-was-successfully&removed)
exten => 1,n,Wait(1)
exten => 1,n,Hangup
;--== end of [app-blacklist-remove] ==--;
[app-blacklist-add]
exten => s,1,Answer
exten => s,n,Macro(user-callerid,)
exten => s,n,Wait(1)
exten => s,n,Set(NumLoops=0)
exten => s,n(start),Playback(enter-num-blacklist)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=60)
exten => s,n,Read(blacknr,then-press-pound,,,,)
exten => s,n,SayDigits(${blacknr})
exten => s,n,Playback(if-correct-press&digits/1)
exten => s,n,Noop(Waiting for input)
exten => s,n(end),WaitExten(60,)
exten => s,n,Playback(sorry-youre-having-problems&goodbye)
exten => 1,1,GotoIf($[ "${blacknr}" != ""]?:app-blacklist-add-invalid,s,1)
exten => 1,n,Set(DB(blacklist/${blacknr})=1)
exten => 1,n,Playback(num-was-successfully&added)
exten => 1,n,Wait(1)
exten => 1,n,Hangup
;--== end of [app-blacklist-add] ==--;
[app-blacklist-add-invalid]
exten => s,1,Set(NumLoops=$[${NumLoops} + 1])
exten => s,n,Playback(pm-invalid-option)
exten => s,n,GotoIf($[${NumLoops} < 3]?app-blacklist-add,s,start)
exten => s,n,Playback(goodbye)
exten => s,n,Hangup
;--== end of [app-blacklist-add-invalid] ==--;
[app-cf-toggle]
exten => *96,1,Answer
exten => *96,n,Wait(1)
exten => *96,n,Macro(user-callerid,)
exten => *96,n,Set(fromext=${AMPUSER})
exten => *96,n,GotoIf($["${DB(CF/${fromext})}" = ""]?activate:deactivate)
exten => *96,n(activate),Read(toext,ent-target-attendant&then-press-pound,,,,)
exten => *96,n,GotoIf($["${toext}"=""]?activate)
exten => *96,n,Wait(1)
exten => *96,n(toext),Set(DB(CF/${fromext})=${toext})
exten => *96,n,Set(STATE=BUSY)
exten => *96,n,Gosub(app-cf-toggle,sstate,1())
exten => *96,n(hook_on),Playback(call-fwd-unconditional&for&extension)
exten => *96,n,SayDigits(${fromext})
exten => *96,n,Playback(is-set-to)
exten => *96,n,SayDigits(${toext})
exten => *96,n,Macro(hangupcall,)
exten => *96,n(setdirect),Answer
exten => *96,n,Wait(1)
exten => *96,n,Macro(user-callerid,)
exten => *96,n,Goto(toext)
exten => *96,n(deactivate),Noop(Deleting: CF/${fromext} ${DB_DELETE(CF/${fromext})})
exten => *96,n,Set(STATE=NOT_INUSE)
exten => *96,n,Gosub(app-cf-toggle,sstate,1())
exten => *96,n(hook_off),Playback(call-fwd-unconditional&de-activated)
exten => *96,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICE_STATE(Custom:CF${fromext})=${STATE})
exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${fromext}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVCF${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-cf-toggle] ==--;
[app-cf-unavailable-prompt-on]
exten => *95,1,Answer
exten => *95,n,Wait(1)
exten => *95,n,Macro(user-callerid,)
exten => *95,n,Read(fromext,call-fwd-no-ans&please-enter-your&extension&then-press-pound,,,,)
exten => *95,n,Set(fromext=${IF($["${fromext}"=""]?${AMPUSER}:${fromext})})
exten => *95,n,Wait(1)
exten => *95,n(startread),Read(toext,ent-target-attendant&then-press-pound,,,,)
exten => *95,n,GotoIf($["${toext}"=""]?startread)
exten => *95,n,Wait(1)
exten => *95,n,Set(DB(CFU/${fromext})=${toext})
exten => *95,n(hook_1),Playback(call-fwd-no-ans&for&extension)
exten => *95,n,SayDigits(${fromext})
exten => *95,n,Playback(is-set-to)
exten => *95,n,SayDigits(${toext})
exten => *95,n,Macro(hangupcall,)
exten => _*95.,1,Answer
exten => _*95.,n,Wait(1)
exten => _*95.,n,Macro(user-callerid,)
exten => _*95.,n,Set(fromext=${AMPUSER})
exten => _*95.,n,Set(toext=${EXTEN:3})
exten => _*95.,n,Set(DB(CFU/${fromext})=${toext})
exten => _*95.,n(hook_2),Playback(call-fwd-no-ans&for&extension)
exten => _*95.,n,SayDigits(${fromext})
exten => _*95.,n,Playback(is-set-to)
exten => _*95.,n,SayDigits(${toext})
exten => _*95.,n,Macro(hangupcall,)
;--== end of [app-cf-unavailable-prompt-on] ==--;
[app-cf-busy-prompting-on]
exten => *94,1,Answer
exten => *94,n,Wait(1)
exten => *94,n,Macro(user-callerid,)
exten => *94,n,Read(fromext,call-fwd-on-busy&please-enter-your&extension&then-press-pound,,,,)
exten => *94,n,Set(fromext=${IF($["${fromext}"=""]?${AMPUSER}:${fromext})})
exten => *94,n,Wait(1)
exten => *94,n(startread),Read(toext,ent-target-attendant&then-press-pound,,,,)
exten => *94,n,GotoIf($["${toext}"=""]?startread)
exten => *94,n,Wait(1)
exten => *94,n,Set(DB(CFB/${fromext})=${toext})
exten => *94,n(hook_1),Playback(call-fwd-on-busy&for&extension)
exten => *94,n,SayDigits(${fromext})
exten => *94,n,Playback(is-set-to)
exten => *94,n,SayDigits(${toext})
exten => *94,n,Macro(hangupcall,)
exten => _*94.,1,Answer
exten => _*94.,n,Wait(1)
exten => _*94.,n,Macro(user-callerid,)
exten => _*94.,n,Set(fromext=${AMPUSER})
exten => _*94.,n,Set(toext=${EXTEN:3})
exten => _*94.,n,Set(DB(CFB/${fromext})=${toext})
exten => _*94.,n(hook_2),Playback(call-fwd-on-busy&for&extension)
exten => _*94.,n,SayDigits(${fromext})
exten => _*94.,n,Playback(is-set-to)
exten => _*94.,n,SayDigits(${toext})
exten => _*94.,n,Macro(hangupcall,)
;--== end of [app-cf-busy-prompting-on] ==--;
[app-cf-prompting-on]
exten => *93,1,Answer
exten => *93,n,Wait(1)
exten => *93,n,Macro(user-callerid,)
exten => *93,n,Read(fromext,call-fwd-unconditional&please-enter-your&extension&then-press-pound,,,,)
exten => *93,n,Set(fromext=${IF($["foo${fromext}"="foo"]?${AMPUSER}:${fromext})})
exten => *93,n,Wait(1)
exten => *93,n(startread),Read(toext,ent-target-attendant&then-press-pound,,,,)
exten => *93,n,GotoIf($["foo${toext}"="foo"]?startread)
exten => *93,n,Wait(1)
exten => *93,n,Set(DB(CF/${fromext})=${toext})
exten => *93,n,Set(STATE=BUSY)
exten => *93,n,Gosub(app-cf-prompting-on,sstate,1())
exten => *93,n(hook_1),Playback(call-fwd-unconditional&for&extension)
exten => *93,n,SayDigits(${fromext})
exten => *93,n,Playback(is-set-to)
exten => *93,n,SayDigits(${toext})
exten => *93,n,Macro(hangupcall,)
exten => _*93.,1,Answer
exten => _*93.,n,Wait(1)
exten => _*93.,n,Macro(user-callerid,)
exten => _*93.,n,Set(fromext=${AMPUSER})
exten => _*93.,n,Set(toext=${EXTEN:3})
exten => _*93.,n,Set(DB(CF/${fromext})=${toext})
exten => _*93.,n,Set(STATE=BUSY)
exten => _*93.,n,Gosub(app-cf-prompting-on,sstate,1())
exten => _*93.,n(hook_2),Playback(call-fwd-unconditional&for&extension)
exten => _*93.,n,SayDigits(${fromext})
exten => _*93.,n,Playback(is-set-to)
exten => _*93.,n,SayDigits(${toext})
exten => _*93.,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICE_STATE(Custom:CF${fromext})=${STATE})
exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${fromext}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVCF${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-cf-prompting-on] ==--;
[app-cf-busy-off-any]
exten => *92,1,Answer
exten => *92,n,Macro(user-callerid,)
exten => *92,n,Wait(1)
exten => *92,n,Read(fromext,please-enter-your&extension&then-press-pound,,,,)
exten => *92,n,Set(fromext=${IF($["foo${fromext}"="foo"]?${AMPUSER}:${fromext})})
exten => *92,n,Wait(1)
exten => *92,n,Noop(Deleting: CFB/${fromext} ${DB_DELETE(CFB/${fromext})})
exten => *92,n(hook_1),Playback(call-fwd-on-busy&for&extension)
exten => *92,n,SayDigits(${fromext})
exten => *92,n,Playback(cancelled)
exten => *92,n,Macro(hangupcall,)
;--== end of [app-cf-busy-off-any] ==--;
[app-cf-busy-off]
exten => *91,1,Answer
exten => *91,n,Wait(1)
exten => *91,n,Macro(user-callerid,)
exten => *91,n,Set(fromext=${AMPUSER})
exten => *91,n,Noop(Deleting: CFB/${fromext} ${DB_DELETE(CFB/${fromext})})
exten => *91,n(hook_1),Playback(call-fwd-on-busy&de-activated)
exten => *91,n,Macro(hangupcall,)
exten => _*91.,1,Answer
exten => _*91.,n,Wait(1)
exten => _*91.,n,Set(fromext=${EXTEN:3})
exten => _*91.,n,Noop(Deleting: CFB/${fromext} ${DB_DELETE(CFB/${fromext})})
exten => _*91.,n(hook_2),Playback(call-fwd-on-busy&for&extension)
exten => _*91.,n,SayDigits(${fromext})
exten => _*91.,n,Playback(cancelled)
exten => _*91.,n,Macro(hangupcall,)
;--== end of [app-cf-busy-off] ==--;
[app-cf-busy-on]
exten => *90,1,Answer
exten => *90,n,Wait(1)
exten => *90,n,Macro(user-callerid,)
exten => *90,n,Set(fromext=${AMPUSER})
exten => *90,n,GotoIf($["${fromext}"!=""]?startread)
exten => *90,n,Playback(agent-loggedoff)
exten => *90,n,Macro(hangupcall,)
exten => *90,n(startread),Read(toext,ent-target-attendant&then-press-pound,,,,)
exten => *90,n,GotoIf($["${toext}"=""]?startread)
exten => *90,n,Wait(1)
exten => *90,n,Set(DB(CFB/${fromext})=${toext})
exten => *90,n(hook_1),Playback(call-fwd-on-busy&for&extension)
exten => *90,n,SayDigits(${fromext})
exten => *90,n,Playback(is-set-to)
exten => *90,n,SayDigits(${toext})
exten => *90,n,Macro(hangupcall,)
exten => _*90.,1,Answer
exten => _*90.,n,Wait(1)
exten => _*90.,n,Macro(user-callerid,)
exten => _*90.,n,Set(fromext=${AMPUSER})
exten => _*90.,n,Set(toext=${EXTEN:3})
exten => _*90.,n,Set(DB(CFB/${fromext})=${toext})
exten => _*90.,n(hook_2),Playback(call-fwd-on-busy&for&extension)
exten => _*90.,n,SayDigits(${fromext})
exten => _*90.,n,Playback(is-set-to)
exten => _*90.,n,SayDigits(${toext})
exten => _*90.,n,Macro(hangupcall,)
;--== end of [app-cf-busy-on] ==--;
[app-cf-off-any]
exten => *74,1,Answer
exten => *74,n,Macro(user-callerid,)
exten => *74,n,Wait(1)
exten => *74,n,Read(fromext,please-enter-your&extension&then-press-pound,,,,)
exten => *74,n,Set(fromext=${IF($["foo${fromext}"="foo"]?${AMPUSER}:${fromext})})
exten => *74,n,Wait(1)
exten => *74,n,Noop(Deleting: CF/${fromext} ${DB_DELETE(CF/${fromext})})
exten => *74,n,Set(STATE=NOT_INUSE)
exten => *74,n,Gosub(app-cf-off-any,sstate,1())
exten => *74,n(hook_1),Playback(call-fwd-unconditional&for&extension)
exten => *74,n,SayDigits(${fromext})
exten => *74,n,Playback(cancelled)
exten => *74,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICE_STATE(Custom:CF${fromext})=${STATE})
exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${fromext}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVCF${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-cf-off-any] ==--;
[app-cf-off]
exten => *73,1,Answer
exten => *73,n,Wait(1)
exten => *73,n,Macro(user-callerid,)
exten => *73,n,Set(fromext=${AMPUSER})
exten => *73,n,Noop(Deleting: CF/${fromext} ${DB_DELETE(CF/${fromext})})
exten => *73,n,Set(STATE=NOT_INUSE)
exten => *73,n,Gosub(app-cf-off,sstate,1())
exten => *73,n(hook_1),Playback(call-fwd-unconditional&de-activated)
exten => *73,n,Macro(hangupcall,)
exten => _*73.,1,Answer
exten => _*73.,n,Wait(1)
exten => _*73.,n,Set(fromext=${EXTEN:3})
exten => _*73.,n,Noop(Deleting: CF/${fromext} ${DB_DELETE(CF/${fromext})})
exten => _*73.,n,Set(STATE=NOT_INUSE)
exten => _*73.,n,Gosub(app-cf-off,sstate,1())
exten => _*73.,n(hook_2),Playback(call-fwd-unconditional&for&extension)
exten => _*73.,n,SayDigits(${fromext})
exten => _*73.,n,Playback(cancelled)
exten => _*73.,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICE_STATE(Custom:CF${fromext})=${STATE})
exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${fromext}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVCF${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-cf-off] ==--;
[app-cf-on]
exten => *72,1,Answer
exten => *72,n,Wait(1)
exten => *72,n,Macro(user-callerid,)
exten => *72,n,Set(fromext=${AMPUSER})
exten => *72,n,GotoIf($["${fromext}"!=""]?startread)
exten => *72,n,Playback(agent-loggedoff)
exten => *72,n,Macro(hangupcall,)
exten => *72,n(startread),Read(toext,ent-target-attendant&then-press-pound,,,,)
exten => *72,n,GotoIf($["foo${toext}"="foo"]?startread)
exten => *72,n,Wait(1)
exten => *72,n,Set(DB(CF/${fromext})=${toext})
exten => *72,n,Set(STATE=BUSY)
exten => *72,n,Gosub(app-cf-on,sstate,1())
exten => *72,n(hook_1),Playback(call-fwd-unconditional&for&extension)
exten => *72,n,SayDigits(${fromext})
exten => *72,n,Playback(is-set-to)
exten => *72,n,SayDigits(${toext})
exten => *72,n,Macro(hangupcall,)
exten => _*72.,1,Answer
exten => _*72.,n,Wait(1)
exten => _*72.,n,Macro(user-callerid,)
exten => _*72.,n,Set(fromext=${AMPUSER})
exten => _*72.,n,Set(toext=${EXTEN:3})
exten => _*72.,n,Set(DB(CF/${fromext})=${toext})
exten => _*72.,n,Set(STATE=BUSY)
exten => _*72.,n,Gosub(app-cf-on,sstate,1())
exten => _*72.,n(hook_2),Playback(call-fwd-unconditional&for&extension)
exten => _*72.,n,SayDigits(${fromext})
exten => _*72.,n,Playback(is-set-to)
exten => _*72.,n,SayDigits(${toext})
exten => _*72.,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICE_STATE(Custom:CF${fromext})=${STATE})
exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${fromext}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVCF${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-cf-on] ==--;
[app-cf-unavailable-off]
exten => *53,1,Answer
exten => *53,n,Wait(1)
exten => *53,n,Macro(user-callerid,)
exten => *53,n,Set(fromext=${AMPUSER})
exten => *53,n,Noop(Deleting: CFU/${fromext} ${DB_DELETE(CFU/${fromext})})
exten => *53,n(hook_1),Playback(call-fwd-no-ans&de-activated)
exten => *53,n,Macro(hangupcall,)
exten => _*53.,1,Answer
exten => _*53.,n,Wait(1)
exten => _*53.,n,Set(fromext=${EXTEN:3})
exten => _*53.,n,Noop(Deleting: CFU/${fromext} ${DB_DELETE(CFU/${fromext})})
exten => _*53.,n(hook_2),Playback(call-fwd-no-ans&for&extension)
exten => _*53.,n,SayDigits(${fromext})
exten => _*53.,n,Playback(cancelled)
exten => _*53.,n,Macro(hangupcall,)
;--== end of [app-cf-unavailable-off] ==--;
[app-cf-unavailable-on]
exten => *52,1,Answer
exten => *52,n,Wait(1)
exten => *52,n,Macro(user-callerid,)
exten => *52,n,Set(fromext=${AMPUSER})
exten => *52,n,GotoIf($["${fromext}"!=""]?startread)
exten => *52,n,Playback(agent-loggedoff)
exten => *52,n,Macro(hangupcall,)
exten => *52,n(startread),Read(toext,ent-target-attendant&then-press-pound,,,,)
exten => *52,n,GotoIf($["${toext}"=""]?startread)
exten => *52,n,Wait(1)
exten => *52,n,Set(DB(CFU/${fromext})=${toext})
exten => *52,n(hook_1),Playback(call-fwd-no-ans&for&extension)
exten => *52,n,SayDigits(${fromext})
exten => *52,n,Playback(is-set-to)
exten => *52,n,SayDigits(${toext})
exten => *52,n,Macro(hangupcall,)
exten => _*52.,1,Answer
exten => _*52.,n,Wait(1)
exten => _*52.,n,Macro(user-callerid,)
exten => _*52.,n,Set(fromext=${AMPUSER})
exten => _*52.,n,Set(toext=${EXTEN:3})
exten => _*52.,n,Set(DB(CFU/${fromext})=${toext})
exten => _*52.,n(hook_2),Playback(call-fwd-no-ans&for&extension)
exten => _*52.,n,SayDigits(${fromext})
exten => _*52.,n,Playback(is-set-to)
exten => _*52.,n,SayDigits(${toext})
exten => _*52.,n,Macro(hangupcall,)
;--== end of [app-cf-unavailable-on] ==--;
[sub-record-cancel]
exten => s,1,Return()
;--== end of [sub-record-cancel] ==--;
[sub-record-check]
exten => s,1,GotoIf($[${LEN(${FROMEXTEN})}]?initialized)
exten => s,n,Set(__REC_STATUS=INITIALIZED)
exten => s,n,Set(NOW=${EPOCH})
exten => s,n,Set(__DAY=${STRFTIME(${NOW},,%d)})
exten => s,n,Set(__MONTH=${STRFTIME(${NOW},,%m)})
exten => s,n,Set(__YEAR=${STRFTIME(${NOW},,%Y)})
exten => s,n,Set(__TIMESTR=${YEAR}${MONTH}${DAY}-${STRFTIME(${NOW},,%H%M%S)})
exten => s,n,Set(__FROMEXTEN=${IF($[${LEN(${AMPUSER})}]?${AMPUSER}:${IF($[${LEN(${REALCALLERIDNUM})}]?${REALCALLERIDNUM}:unknown)})})
exten => s,n,Set(__MON_FMT=${IF($["${MIXMON_FORMAT}"="wav49"]?WAV:${MIXMON_FORMAT})})
exten => s,n(initialized),Noop(Recordings initialized)
exten => s,n,ExecIf($[!${LEN(${ARG3})}]?Set(ARG3=dontcare))
exten => s,n,Set(REC_POLICY_MODE_SAVE=${REC_POLICY_MODE})
exten => s,n,ExecIf($["${BLINDTRANSFER}${ATTENDEDTRANSFER}" != ""]?Set(REC_STATUS=NO))
exten => s,n(next),GotoIf($[${LEN(${ARG1})}]?checkaction)
exten => s,n(recorderror),Playback(something-terribly-wrong,error)
exten => s,n,Hangup
exten => s,n(checkaction),GotoIf($[${DIALPLAN_EXISTS(sub-record-check,${ARG1})}]?sub-record-check,${ARG1},1)
exten => s,n,Noop(Generic ${ARG1} Recording Check - ${FROMEXTEN} ${ARG2})
exten => s,n,Gosub(recordcheck,1(${ARG3},${ARG1},${ARG2}))
exten => s,n,Return()
exten => recordcheck,1,Noop(Starting recording check against ${ARG1})
exten => recordcheck,n,Goto(${ARG1})
exten => recordcheck,n(dontcare),Return()
exten => recordcheck,n(always),Noop(Detected legacy "always" entry. Mapping to "force")
exten => recordcheck,n(force),Set(__REC_POLICY_MODE=FORCE)
exten => recordcheck,n,GotoIf($["${REC_STATUS}"!="RECORDING"]?startrec)
exten => recordcheck,n,Return()
exten => recordcheck,n(delayed),Noop(Detected legacy "delayed" entry. Mapping to "yes")
exten => recordcheck,n(yes),ExecIf($["${REC_POLICY_MODE}" = "NEVER" | "${REC_POLICY_MODE}" = "NO" | "${REC_STATUS}" = "RECORDING"]?Return())
exten => recordcheck,n,Set(__REC_POLICY_MODE=YES)
exten => recordcheck,n,Goto(startrec)
exten => recordcheck,n(no),Set(__REC_POLICY_MODE=NO)
exten => recordcheck,n,Return()
exten => recordcheck,n(never),Set(__REC_POLICY_MODE=NEVER)
exten => recordcheck,n,Goto(stoprec)
exten => recordcheck,n(startrec),Noop(Starting recording: ${ARG2}, ${ARG3})
exten => recordcheck,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => recordcheck,n,Set(__CALLFILENAME=${ARG2}-${ARG3}-${FROMEXTEN}-${TIMESTR}-${UNIQUEID})
exten => recordcheck,n,MixMonitor(${MIXMON_DIR}${YEAR}/${MONTH}/${DAY}/${CALLFILENAME}.${MON_FMT},ai(LOCAL_MIXMON_ID),${MIXMON_POST})
exten => recordcheck,n,Set(__MIXMON_ID=${LOCAL_MIXMON_ID})
exten => recordcheck,n,Set(__RECORD_ID=${CHANNEL(name)})
exten => recordcheck,n,Set(__REC_STATUS=RECORDING)
exten => recordcheck,n,Set(CDR(recordingfile)=${CALLFILENAME}.${MON_FMT})
exten => recordcheck,n,Return()
exten => recordcheck,n(stoprec),Noop(Stopping recording: ${ARG2}, ${ARG3})
exten => recordcheck,n,Set(__REC_STATUS=STOPPED)
exten => recordcheck,n,System(/var/lib/asterisk/bin/stoprecording.php "${CHANNEL(name)}")
exten => recordcheck,n,Return()
exten => out,1,Noop(Outbound Recording Check from ${FROMEXTEN} to ${ARG2})
exten => out,n,Set(RECMODE=${DB(AMPUSER/${FROMEXTEN}/recording/out/external)})
exten => out,n,ExecIf($[!${LEN(${RECMODE})} | "${RECMODE}" = "dontcare"]?Goto(routewins))
exten => out,n,ExecIf($["${ARG3}" = "never" | "${ARG3}" = "force"]?Goto(routewins))
exten => out,n(extenwins),Gosub(recordcheck,1(${RECMODE},out,${ARG2}))
exten => out,n,Return()
exten => out,n(routewins),Gosub(recordcheck,1(${ARG3},out,${ARG2}))
exten => out,n,Return()
exten => in,1,Noop(Inbound Recording Check to ${ARG2})
exten => in,n,Set(FROMEXTEN=unknown)
exten => in,n,ExecIf($[${LEN(${CALLERID(num)})}]?Set(FROMEXTEN=${CALLERID(num)}))
exten => in,n,Gosub(recordcheck,1(${ARG3},in,${ARG2}))
exten => in,n,Return()
exten => exten,1,Noop(Exten Recording Check between ${FROMEXTEN} and ${ARG2})
exten => exten,n,Set(CALLTYPE=${IF($[${LEN(${FROM_DID})}]?external:internal)})
exten => exten,n,ExecIf(${LEN(${CALLTYPE_OVERRIDE})}?Set(CALLTYPE=${CALLTYPE_OVERRIDE}))
exten => exten,n,Set(CALLEE=${DB(AMPUSER/${ARG2}/recording/in/${CALLTYPE})})
exten => exten,n,ExecIf($[!${LEN(${CALLEE})}]?Set(CALLEE=dontcare))
exten => exten,n,GotoIf($["${CALLTYPE}"="external"]?callee)
exten => exten,n,GotoIf($["${CALLEE}"="dontcare"]?caller)
exten => exten,n,ExecIf($[${LEN(${DB(AMPUSER/${FROMEXTEN}/recording/priority)})}]?Set(CALLER_PRI=${DB(AMPUSER/${FROMEXTEN}/recording/priority)}):Set(CALLER_PRI=0))
exten => exten,n,ExecIf($[${LEN(${DB(AMPUSER/${ARG2}/recording/priority)})}]?Set(CALLEE_PRI=${DB(AMPUSER/${ARG2}/recording/priority)}):Set(CALLEE_PRI=0))
exten => exten,n,GotoIf($["${CALLER_PRI}"="${CALLEE_PRI}"]?${REC_POLICY}:${IF($[${CALLER_PRI}>${CALLEE_PRI}]?caller:callee)})
exten => exten,n(callee),Gosub(recordcheck,1(${CALLEE},${CALLTYPE},${ARG2}))
exten => exten,n,Return()
exten => exten,n(caller),Set(RECMODE=${DB(AMPUSER/${FROMEXTEN}/recording/out/internal)})
exten => exten,n,ExecIf($[!${LEN(${RECMODE})}]?Set(RECMODE=dontcare))
exten => exten,n,ExecIf($["${RECMODE}"="dontcare"]?Set(RECMODE=${CALLEE}))
exten => exten,n,Gosub(recordcheck,1(${RECMODE},${CALLTYPE},${ARG2}))
exten => exten,n,Return()
exten => conf,1,Noop(Conference Recording Check ${FROMEXTEN} to ${ARG2})
exten => conf,n,Gosub(recconf,1(${ARG2},${ARG2},${ARG3}))
exten => conf,n,Return()
exten => page,1,Noop(Paging Recording Check ${FROMEXTEN} to ${ARG2})
exten => page,n,GosubIf($["${REC_POLICY_MODE}"="always"]?recconf,1(${ARG2},${FROMEXTEN},${ARG3}))
exten => page,n,Return()
exten => recconf,1,Noop(Setting up recording: ${ARG1}, ${ARG2}, ${ARG3})
exten => recconf,n,Set(__CALLFILENAME=${IF($[${CONFBRIDGE_INFO(parties,${ARG2})}]?${DB(RECCONF/${ARG2})}:${ARG1}-${ARG2}-${ARG3}-${TIMESTR}-${UNIQUEID})})
exten => recconf,n,ExecIf($[!${CONFBRIDGE_INFO(parties,${ARG2})}]?Set(DB(RECCONF/${ARG2})=${CALLFILENAME}))
exten => recconf,n,Set(CONFBRIDGE(bridge,record_file)=${MIXMON_DIR}${YEAR}/${MONTH}/${DAY}/${CALLFILENAME}.${MON_FMT})
exten => recconf,n,ExecIf($["${ARG3}"!="always"]?Return())
exten => recconf,n,Set(CONFBRIDGE(bridge,record_conference)=yes)
exten => recconf,n,Set(__REC_STATUS=RECORDING)
exten => recconf,n,Set(CDR(recordingfile)=${CALLFILENAME}.${MON_FMT})
exten => recconf,n,Return()
exten => recq,1,Noop(Setting up recording: ${ARG1}, ${ARG2}, ${ARG3})
exten => recq,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => recq,n,Set(MONITOR_FILENAME=${MIXMON_DIR}${YEAR}/${MONTH}/${DAY}/${CALLFILENAME})
exten => recq,n,MixMonitor(${MONITOR_FILENAME}.${MON_FMT},${MONITOR_OPTIONS},${MIXMON_POST})
exten => recq,n,Set(__REC_STATUS=RECORDING)
exten => recq,n,Set(CDR(recordingfile)=${CALLFILENAME}.${MON_FMT})
exten => recq,n,Return()
exten => parking,1,Noop(User ${ARG2} picked up a parked call)
exten => parking,n,Set(USER=${ARG2})
exten => parking,n,ExecIf($[!${LEN(${ARG2})}]?Set(USER=unknown))
exten => parking,n,Set(RECMODE=${DB(AMPUSER/${ARG2}/recording/out/internal)})
exten => parking,n,ExecIf($[!${LEN(${RECMODE})}]?Set(RECMODE=dontcare))
exten => parking,n,Gosub(recordcheck,1(${RECMODE},parked,${USER}))
exten => parking,n,Return()
;--== end of [sub-record-check] ==--;
[macro-one-touch-record]
exten => s,1,Set(ONETOUCH_REC_SCRIPT_STATUS=)
exten => s,n,System(/var/lib/asterisk/bin/one_touch_record.php "${CHANNEL(name)}")
exten => s,n,Noop(ONETOUCH_REC_SCRIPT_STATUS: [${ONETOUCH_REC_SCRIPT_STATUS}])
exten => s,n,ExecIf($["${REC_STATUS}"="RECORDING"]?Playback(beep))
exten => s,n,ExecIf($["${REC_STATUS}"="STOPPED"]?Playback(beep&beep))
exten => s,n,ExecIf($["${ONETOUCH_REC_SCRIPT_STATUS:0:6}"="DENIED"]?Playback(access-denied))
exten => s,n,MacroExit()
;--== end of [macro-one-touch-record] ==--;
[app-callwaiting-cwoff]
exten => *71,1,Answer
exten => *71,n,Wait(1)
exten => *71,n,Macro(user-callerid,)
exten => *71,n,Noop(Deleting: CW/${AMPUSER} ${DB_DELETE(CW/${AMPUSER})})
exten => *71,n(hook_1),Playback(call-waiting&de-activated)
exten => *71,n,Macro(hangupcall,)
;--== end of [app-callwaiting-cwoff] ==--;
[app-callwaiting-cwon]
exten => *70,1,Answer
exten => *70,n,Wait(1)
exten => *70,n,Macro(user-callerid,)
exten => *70,n,Set(DB(CW/${AMPUSER})=ENABLED)
exten => *70,n(hook_1),Playback(call-waiting&activated)
exten => *70,n,Macro(hangupcall,)
;--== end of [app-callwaiting-cwon] ==--;
[app-campon-toggle]
exten => *84,1,Answer
exten => *84,n,Set(CCSS_SETUP=TRUE)
exten => *84,n,Macro(user-callerid,)
exten => *84,n,ExecIf($["${EXTENSION_STATE(*84${AMPUSER}@ext-local)}" = "INUSE"]?CallCompletionCancel():CallCompletionRequest())
exten => *84,n,Playback(beep)
exten => *84,n,Macro(hangupcall,)
;--== end of [app-campon-toggle] ==--;
[ext-local]
exten => *84300,1,Goto(app-campon-toggle,*84,1)
exten => *84300,hint,ccss:PJSIP/300
exten => *300,1,Macro(vm,300,DIRECTDIAL,${IVR_RETVM})
exten => *300,n,Goto(vmret,1)
exten => 300,1,Set(__RINGTIMER=${IF($["${DB(AMPUSER/300/ringtimer)}" > "0"]?${DB(AMPUSER/300/ringtimer)}:${RINGTIMER_DEFAULT})})
exten => 300,n,Macro(exten-vm,300,300,0,0,0)
exten => 300,n(dest),Set(__PICKUPMARK=)
exten => 300,n,Macro(vm,300,${DIALSTATUS},${IVR_RETVM})
exten => 300,n,Goto(vmret,1)
exten => 300,hint,PJSIP/300&Custom:DND300,CustomPresence:300
exten => vmb300,1,Macro(vm,300,BUSY,${IVR_RETVM})
exten => vmb300,n,Goto(vmret,1)
exten => vmu300,1,Macro(vm,300,NOANSWER,${IVR_RETVM})
exten => vmu300,n,Goto(vmret,1)
exten => vms300,1,Macro(vm,300,NOMESSAGE,${IVR_RETVM})
exten => vms300,n,Goto(vmret,1)
exten => vmi300,1,Macro(vm,300,INSTRUCT,${IVR_RETVM})
exten => vmi300,n,Goto(vmret,1)
exten => vmret,1,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => vmret,n,Hangup
exten => vmret,n(playret),Playback(exited-vm-will-be-transfered&silence/1)
exten => vmret,n,Goto(${IVR_CONTEXT},return,1)
exten => h,1,Macro(hangupcall,)
;--== end of [ext-local] ==--;
[app-campon-cancel]
exten => *83,1,Answer
exten => *83,n,Set(CCSS_SETUP=TRUE)
exten => *83,n,Macro(user-callerid,)
exten => *83,n,CallCompletionCancel()
exten => *83,n,Playback(beep)
exten => *83,n,Macro(hangupcall,)
;--== end of [app-campon-cancel] ==--;
[app-campon-request]
exten => *82,1,Answer
exten => *82,n,Set(CCSS_SETUP=TRUE)
exten => *82,n,Macro(user-callerid,)
exten => *82,n,CallCompletionRequest()
exten => *82,n,Playback(beep)
exten => *82,n,Macro(hangupcall,)
;--== end of [app-campon-request] ==--;
[sub-ccss]
exten => s,1,ExecIf($[${LEN(${CCSS_SETUP})}]?Return())
exten => s,n,Set(CCSS_SETUP=TRUE)
exten => s,n(monitor),GosubIf($[${LEN(${DB(AMPUSER/${ARG2}/ccss/cc_monitor_policy)})}]?monitor_config,1(${ARG1},${ARG2}):monitor_default,1(${ARG1},${ARG2}))
exten => s,n(agent),GosubIf($[${LEN(${DB(AMPUSER/${AMPUSER}/ccss/cc_agent_policy)})}]?agent_config,1():agent_default,1())
exten => s,n,Set(DB(AMPUSER/${AMPUSER}/ccss/last_number)=${ARG2})
exten => s,n,Return(${GOSUB_RETVAL})
exten => monitor_config,1,Set(CALLCOMPLETION(cc_monitor_policy)=${DB(AMPUSER/${ARG2}/ccss/cc_monitor_policy)})
exten => monitor_config,n,GotoIf($["CALLCOMPLETION(cc_monitor_policy)" != "never"]?set_monitor)
exten => monitor_config,n,StackPop()
exten => monitor_config,n,Return(FALSE)
exten => monitor_config,n(set_monitor),Set(CALLCOMPLETION(cc_max_monitors)=${DB(AMPUSER/${ARG2}/ccss/max_monitors)})
exten => monitor_config,n,Return(TRUE)
exten => monitor_default,1,GotoIf($["${DB(AMPUSER/${ARG2}/cidname)}" != ""]?is_exten)
exten => monitor_default,n,StackPop()
exten => monitor_default,n,Return(FALSE)
exten => monitor_default,n(is_exten),Set(CALLCOMPLETION(cc_monitor_policy)=generic)
exten => monitor_default,n,Set(CALLCOMPLETION(cc_max_monitors)=5)
exten => monitor_default,n,Return(TRUE)
exten => agent_config,1,Set(CALLCOMPLETION(cc_agent_policy)=${DB(AMPUSER/${AMPUSER}/ccss/cc_agent_policy)})
exten => agent_config,n,Set(CALLCOMPLETION(cc_offer_timer)=${DB(AMPUSER/${AMPUSER}/ccss/cc_offer_timer)})
exten => agent_config,n,Set(CALLCOMPLETION(ccbs_available_timer)=${DB(AMPUSER/${AMPUSER}/ccss/ccbs_available_timer)})
exten => agent_config,n,Set(CALLCOMPLETION(ccnr_available_timer)=${DB(AMPUSER/${AMPUSER}/ccss/ccnr_available_timer)})
exten => agent_config,n,Set(CALLCOMPLETION(cc_callback_macro)=ccss-default)
exten => agent_config,n,ExecIf($["${CALLCOMPLETION(cc_agent_policy)}" = "generic"]?Set(CALLCOMPLETION(cc_recall_timer)=${DB(AMPUSER/${AMPUSER}/ccss/cc_recall_timer)}))
exten => agent_config,n,ExecIf($["${CALLCOMPLETION(cc_agent_policy)}" = "generic"]?Set(CALLCOMPLETION(cc_max_agents)=${DB(AMPUSER/${AMPUSER}/ccss/cc_max_agents)}))
exten => agent_config,n,ExecIf($["${DB(AMPUSER/${AMPUSER}/ccss/cc_agent_dialstring)}" != ""]?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/${AMPUSER}_${CALLERID(dnid)}@from-ccss-${DB(AMPUSER/${AMPUSER}/ccss/cc_agent_dialstring)}))
exten => agent_config,n,Set(CALLCOMPLETION(cc_callback_macro)=ccss-default)
exten => agent_config,n,Return()
exten => agent_default,1,Set(CALLCOMPLETION(cc_agent_policy)=generic)
exten => agent_default,n,Set(CALLCOMPLETION(cc_offer_timer)=30)
exten => agent_default,n,Set(CALLCOMPLETION(ccbs_available_timer)=4800)
exten => agent_default,n,Set(CALLCOMPLETION(ccnr_available_timer)=7200)
exten => agent_default,n,Set(CALLCOMPLETION(cc_callback_macro)=ccss-default)
exten => agent_default,n,Set(CALLCOMPLETION(cc_recall_timer)=15)
exten => agent_default,n,Set(CALLCOMPLETION(cc_agent_dialstring)=Local/${AMPUSER}_${CALLERID(dnid)}@from-ccss-extension)
exten => agent_default,n,Return(TRUE)
;--== end of [sub-ccss] ==--;
[macro-ccss-default]
exten => s,1,Playback(beep&calling&extension)
exten => s,n,SayDigits(${CC_EXTEN})
exten => s,n(siprm),SIPRemoveHeader(${CC_HEADER})
exten => s,n,Set(CALLERID(dnid)=${CC_EXTEN})
exten => s,n,Set(CALLERID(name)=${CALLERID(name)})
exten => s,n,MacroExit()
;--== end of [macro-ccss-default] ==--;
[from-ccss-internal]
exten => _X._X.,1,Gosub(sub-from-ccss,s,1(${CUT(EXTEN,_,2)}))
exten => _X._X.,n(no_alert),Goto(from-internal,${CUT(EXTEN,_,1)},1)
;--== end of [from-ccss-internal] ==--;
[from-ccss-extension]
exten => _X._X.,1,Gosub(sub-from-ccss,s,1(${CUT(EXTEN,_,2)}))
exten => _X._X.,n,Goto(ext-local,${CUT(EXTEN,_,1)},1)
;--== end of [from-ccss-extension] ==--;
[sub-from-ccss]
exten => s,1,Macro(blkvm-set,)
exten => s,n,Set(__CC_RECALL=1)
exten => s,n,Set(CALLERID(name)=${IF($[${LEN(${DB(AMPUSER/${ARG1}/cidname)})}]?${DB(AMPUSER/${ARG1}/cidname)}:CALLBACK)})
exten => s,n,Set(CALLERID(number)=${ARG1})
exten => s,n,Return()
exten => s,n,SIPAddHeader(Alert-Info: ${CC_HEADER})
exten => s,n,Return()
;--== end of [sub-from-ccss] ==--;
[cidlookup]
exten => cidlookup_2,1,Set(CURLOPT(httptimeout)=7)
exten => cidlookup_2,n,Set(CALLERID(name)=${CURL(https://api.opencnam.com/v2/phone/${CALLERID(num)}?format=pbx&ref=freepbx)})
exten => cidlookup_2,n,Set(current_hour=${STRFTIME(,,%Y-%m-%d %H)})
exten => cidlookup_2,n,Set(last_query_hour=${DB(cidlookup/opencnam_last_query_hour)})
exten => cidlookup_2,n,Set(total_hourly_queries=${DB(cidlookup/opencnam_total_hourly_queries)})
exten => cidlookup_2,n,ExecIf($["${last_query_hour}" != "${current_hour}"]?Set(DB(cidlookup/opencnam_total_hourly_queries)=0))
exten => cidlookup_2,n,ExecIf($["${total_hourly_queries}" = ""]?Set(DB(cidlookup/opencnam_total_hourly_queries)=0))
exten => cidlookup_2,n,Set(DB(cidlookup/opencnam_total_hourly_queries)=${MATH(${DB(cidlookup/opencnam_total_hourly_queries)}+1,i)})
exten => cidlookup_2,n,ExecIf($[${DB(cidlookup/opencnam_total_hourly_queries)} >= 60]?System(${ASTVARLIBDIR}/bin/opencnam-alert.php))
exten => cidlookup_2,n,Set(DB(cidlookup/opencnam_last_query_hour)=${current_hour})
exten => cidlookup_2,n,Return()
exten => cidlookup_return,1,ExecIf($["${DB(cidname/${CALLERID(num)})}" != ""]?Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}))
exten => cidlookup_return,n,Return()
;--== end of [cidlookup] ==--;
[c-x-3-operator-panel-hold]
exten => 432111,1,MusicOnHold(${XMLNamespaceMusicOnHoldClass})
exten => 432111,n,Hangup
;--== end of [c-x-3-operator-panel-hold] ==--;
[c-x-3-operator-panel-voice-mail]
exten => 432112,1,VoiceMail(${XMLNamespaceVoiceMailBox}@${XMLNamespaceVoiceMailBoxContext},u)
exten => 432112,n,Hangup
;--== end of [c-x-3-operator-panel-voice-mail] ==--;
[c-x-3-operator-panel-meetme]
exten => 432113,1,ConfBridge(${XMLNamespaceMeetMeRoomNumber},${XMLNamespaceMeetMeRoomOptions},,)
exten => 432113,n,Hangup
;--== end of [c-x-3-operator-panel-meetme] ==--;
[c-x-3-operator-panel-confbridge]
exten => 432113,1,ConfBridge(${XMLNamespaceMeetMeRoomNumber},,,)
exten => 432113,n,Hangup
;--== end of [c-x-3-operator-panel-confbridge] ==--;
[c-x-3-operator-panel-park]
exten => 432114,1,ParkAndAnnounce(pbx-transfer:PARKED,200,Local/432116@c-x-3-operator-panel-park-announce-answer,${XMLNamespaceParkContext},${XMLNamespaceParkExtension},1)
exten => 432114,n,Hangup
;--== end of [c-x-3-operator-panel-park] ==--;
[c-x-3-operator-panel-park-announce-answer]
exten => 432116,1,Answer
exten => 432116,n,Hangup
;--== end of [c-x-3-operator-panel-park-announce-answer] ==--;
[c-x-3-operator-panel-listen-to-voice-mail]
exten => 432115,1,ControlPlayback(${XMLNamespaceVoiceMailPath},1000,*,#,7,8,9)
exten => 432115,n,Hangup
;--== end of [c-x-3-operator-panel-listen-to-voice-mail] ==--;
[c-x-3-operator-panel-listen-to-recording]
exten => 432118,1,ControlPlayback(${XMLNamespaceRecordingPath},1000,*,#,7,8,9)
exten => 432118,n,Hangup
;--== end of [c-x-3-operator-panel-listen-to-recording] ==--;
[c-x-3-operator-panel-spy]
exten => 432117,1,ChanSpy(${XMLNamespaceChanSpyChannel},${XMLNamespaceChanSpyOptions})
exten => 432117,n,Hangup
;--== end of [c-x-3-operator-panel-spy] ==--;
[c-x-3-operator-panel-pjsip-auto-answer-headers]
exten => addheader,1,Set(PJSIP_HEADER(add,Alert-Info)=<http://www.notused.com>\;info=alert-autoanswer\;delay=0)
exten => addheader,n,Set(PJSIP_HEADER(add,Alert-Info)=Ring Answer)
exten => addheader,n,Set(PJSIP_HEADER(add,Alert-Info)=ring-answer)
exten => addheader,n,Set(PJSIP_HEADER(add,Call-Info)=\;answer-after=0)
;--== end of [c-x-3-operator-panel-pjsip-auto-answer-headers] ==--;
[c-x-3-operator-panel-pjsip-auto-answer-redirect]
exten => _X!,1,ExecIf($["${D_OPTIONS}"==""]?Set(D_OPTIONS=TtrI))
exten => _X!,n,Dial(${CX_AUTOANSWER_REDIRECT_PEER},,${D_OPTIONS}b(c-x-3-operator-panel-pjsip-auto-answer-headers^addheader^1))
;--== end of [c-x-3-operator-panel-pjsip-auto-answer-redirect] ==--;
[app-dictate-send]
exten => *35,1,Answer
exten => *35,n,Macro(user-callerid,)
exten => *35,n,Noop(CallerID is ${AMPUSER})
exten => *35,n,Set(DICTENABLED=${DB(AMPUSER/${AMPUSER}/dictate/enabled)})
exten => *35,n,GotoIf($[$["x${DICTENABLED}"="x"]|$["x${DICTENABLED}"="xdisabled"]]?nodict:dictok)
exten => *35,n(nodict),Playback(feature-not-avail-line)
exten => *35,n,Hangup
exten => *35,n(dictok),Read(DICTFILE,enter-filename-short,,,,)
exten => *35,n,Set(DICTEMAIL=${DB(AMPUSER/${AMPUSER}/dictate/email)})
exten => *35,n,Set(DICTFMT=${DB(AMPUSER/${AMPUSER}/dictate/format)})
exten => *35,n,Set(NAME=${DB(AMPUSER/${AMPUSER}/cidname)})
exten => *35,n,Playback(dictation-being-processed)
exten => *35,n,System(/var/lib/asterisk/bin/audio-email.pl --file /var/lib/asterisk/sounds/dictate/${AMPUSER}/${DICTFILE}.raw --attachment dict-${DICTFILE} --format ${DICTFMT} --to ${DICTEMAIL} --subject "Dictation from ${NAME} Attached")
exten => *35,n,Playback(dictation-sent)
exten => *35,n,Macro(hangupcall,)
;--== end of [app-dictate-send] ==--;
[app-dictate-record]
exten => *34,1,Answer
exten => *34,n,Macro(user-callerid,)
exten => *34,n,Noop(CallerID is ${AMPUSER})
exten => *34,n,Set(DICTENABLED=${DB(AMPUSER/${AMPUSER}/dictate/enabled)})
exten => *34,n,GotoIf($[$["x${DICTENABLED}"="x"]|$["x${DICTENABLED}"="xdisabled"]]?nodict:dictok)
exten => *34,n(nodict),Playback(feature-not-avail-line)
exten => *34,n,Hangup
exten => *34,n(dictok),Dictate(/var/lib/asterisk/sounds/dictate/${AMPUSER})
exten => *34,n,Macro(hangupcall,)
;--== end of [app-dictate-record] ==--;
[app-dnd-off]
exten => *79,1,Answer
exten => *79,n,Wait(1)
exten => *79,n,Macro(user-callerid,)
exten => *79,n,Noop(Deleting: DND/${AMPUSER} ${DB_DELETE(DND/${AMPUSER})})
exten => *79,n,Set(STATE=NOT_INUSE)
exten => *79,n,Gosub(app-dnd-off,sstate,1())
exten => *79,n(hook_1),Playback(do-not-disturb&de-activated)
exten => *79,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICE_STATE(Custom:DND${AMPUSER})=${STATE})
exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${AMPUSER}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVDND${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-dnd-off] ==--;
[app-dnd-on]
exten => *78,1,Answer
exten => *78,n,Wait(1)
exten => *78,n,Macro(user-callerid,)
exten => *78,n,Set(DB(DND/${AMPUSER})=YES)
exten => *78,n,Set(STATE=BUSY)
exten => *78,n,Gosub(app-dnd-on,sstate,1())
exten => *78,n(hook_1),Playback(do-not-disturb&activated)
exten => *78,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICE_STATE(Custom:DND${AMPUSER})=${STATE})
exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${AMPUSER}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVDND${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-dnd-on] ==--;
[app-dnd-toggle]
exten => *76,1,Answer
exten => *76,n,Wait(1)
exten => *76,n,Macro(user-callerid,)
exten => *76,n,GotoIf($["${DB(DND/${AMPUSER})}" = ""]?activate:deactivate)
exten => *76,n(activate),Set(DB(DND/${AMPUSER})=YES)
exten => *76,n,Set(STATE=BUSY)
exten => *76,n,Gosub(app-dnd-toggle,sstate,1())
exten => *76,n(hook_on),Playback(do-not-disturb&activated)
exten => *76,n,Macro(hangupcall,)
exten => *76,n(deactivate),Noop(Deleting: DND/${AMPUSER} ${DB_DELETE(DND/${AMPUSER})})
exten => *76,n,Set(STATE=NOT_INUSE)
exten => *76,n,Gosub(app-dnd-toggle,sstate,1())
exten => *76,n(hook_off),Playback(do-not-disturb&de-activated)
exten => *76,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICE_STATE(Custom:DND${AMPUSER})=${STATE})
exten => sstate,n,Set(DEVICES=${DB(AMPUSER/${AMPUSER}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:DEVDND${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-dnd-toggle] ==--;
[ext-dnd-hints]
exten => *76300,1,Goto(app-dnd-toggle,*76,1)
exten => *76300,hint,Custom:DEVDND300
;--== end of [ext-dnd-hints] ==--;
[ext-fax]
exten => s,1,Macro(user-callerid,)
exten => s,n,Noop(Receiving Fax for: ${FAX_RX_EMAIL} , From: ${CALLERID(all)})
exten => s,n(receivefax),StopPlaytones
exten => s,n,ReceiveFAX(${ASTSPOOLDIR}/fax/${UNIQUEID}.tif,f)
exten => s,n,ExecIf($["${FAXSTATUS:0:6}"="FAILED" && "${FAXERROR}"!="INIT_ERROR"]?Set(FAXSTATUS="FAILED: error: ${FAXERROR} statusstr: ${FAXOPT(statusstr)}"))
exten => s,n,Hangup
exten => h,1,GotoIf($[${STAT(e,${ASTSPOOLDIR}/fax/${UNIQUEID}.tif)} = 0]?failed)
exten => h,n(process),GotoIf($[${LEN(${FAX_RX_EMAIL})} = 0]?noemail)
exten => h,n(delete_opt),Set(DELETE_AFTER_SEND=true)
exten => h,n,System(${ASTVARLIBDIR}/bin/fax2mail.php --to "${FAX_RX_EMAIL}" --dest "${FROM_DID}" --callerid '${STRREPLACE(CALLERID(all),',\\')}' --file ${ASTSPOOLDIR}/fax/${UNIQUEID}.tif --exten "${FAX_FOR}" --delete "${DELETE_AFTER_SEND}" --attachformat "${FAX_ATTACH_FORMAT}")
exten => h,n(end),Macro(hangupcall,)
exten => h,n(noemail),Noop(ERROR: No Email Address to send FAX: status: [${FAXSTATUS}], From: [${CALLERID(all)}])
exten => h,n,Macro(hangupcall,)
exten => h,process+101(failed),Noop(FAX ${FAXSTATUS} for: ${FAX_RX_EMAIL} , From: ${CALLERID(all)})
exten => h,n,Macro(hangupcall,)
;--== end of [ext-fax] ==--;
[app-fax]
exten => 666,1,Set(FAX_RX_EMAIL=)
exten => 666,n,Goto(ext-fax,s,1)
exten => h,1,Macro(hangupcall,)
;--== end of [app-fax] ==--;
[ext-did-0001]
exten => fax,1,Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)})
;--== end of [ext-did-0001] ==--;
[ext-did-0002]
exten => fax,1,Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)})
exten => 04074041501,1,Set(__DIRECTION=INBOUND)
exten => 04074041501,n,Gosub(sub-record-check,s,1(in,${EXTEN},dontcare))
exten => 04074041501,n,Gosub(app-blacklist-check,s,1())
exten => 04074041501,n,Set(__FROM_DID=${EXTEN})
exten => 04074041501,n,Set(CDR(did)=${FROM_DID})
exten => 04074041501,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 04074041501,n,Set(CHANNEL(musicclass)=default)
exten => 04074041501,n,Set(__MOHCLASS=default)
exten => 04074041501,n,GotoIf($["${__REVERSAL_REJECT}"="TRUE" & "${CHANNEL(reversecharge)}"="1" ]?macro-hangupcall)
exten => 04074041501,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 04074041501,n,Set(CALLERPRES()=allowed_not_screened)
exten => 04074041501,n(dest-ext),Goto(from-did-direct,300,1)
;--== end of [ext-did-0002] ==--;
[ext-featurecodes]
exten => *30,1,Goto(from-internal,*30,1)
exten => *31,1,Goto(from-internal,*31,1)
exten => 555,1,Goto(from-internal,555,1)
exten => 7777,1,Goto(from-internal,7777,1)
exten => 888,1,Goto(from-internal,888,1)
exten => 666,1,Goto(from-internal,666,1)
exten => *43,1,Goto(from-internal,*43,1)
exten => *60,1,Goto(from-internal,*60,1)
exten => *85,1,Goto(from-internal,*85,1)
exten => *86,1,Goto(from-internal,*86,1)
exten => 411,1,Goto(from-internal,411,1)
exten => *98,1,Goto(from-internal,*98,1)
;--== end of [ext-featurecodes] ==--;
[app-fmf-toggle]
exten => *21,1,Goto(app-fmf-toggle,s,start)
exten => s,1(start),Answer
exten => s,n,Wait(1)
exten => s,n,Macro(user-callerid,)
exten => s,n,GotoIf($["${DB(AMPUSER/${AMPUSER}/followme/ddial)}" = "EXTENSION"]?activate)
exten => s,n,GotoIf($["${DB(AMPUSER/${AMPUSER}/followme/ddial)}" = "DIRECT"]?deactivate:end)
exten => s,n(deactivate),Set(DB(AMPUSER/${AMPUSER}/followme/ddial)=EXTENSION)
exten => s,n,Set(STATE=NOT_INUSE)
exten => s,n,Gosub(app-fmf-toggle,sstate,1())
exten => s,n(hook_off),Playback(followme&de-activated)
exten => s,n(end),Macro(hangupcall,)
exten => s,n(activate),Set(DB(AMPUSER/${AMPUSER}/followme/ddial)=DIRECT)
exten => s,n,Set(STATE=INUSE)
exten => s,n,Gosub(app-fmf-toggle,sstate,1())
exten => s,n(hook_on),Playback(followme&activated)
exten => s,n,Macro(hangupcall,)
exten => sstate,1,Set(DEVICES=${DB(AMPUSER/${AMPUSER}/device)})
exten => sstate,n,GotoIf($["${DEVICES}" = "" ]?return)
exten => sstate,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => sstate,n,Set(ITER=1)
exten => sstate,n(begin),Set(DEVICE_STATE(Custom:FOLLOWME${CUT(DEVICES,&,${ITER})})=${STATE})
exten => sstate,n,Set(ITER=$[${ITER} + 1])
exten => sstate,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => sstate,n(return),Return()
;--== end of [app-fmf-toggle] ==--;
[ext-findmefollow]
exten => FM300,1,Goto(300,FM300)
exten => 300,1,GotoIf($[ "${DB(AMPUSER/300/followme/ddial)}" = "EXTENSION" ]?ext-local,300,1)
exten => 300,n(FM300),Macro(user-callerid,)
exten => 300,n,Set(DIAL_OPTIONS=${DIAL_OPTIONS}I)
exten => 300,n,Set(CONNECTEDLINE(num,i)=300)
exten => 300,n,Gosub(sub-presencestate-display,s,1(300))
exten => 300,n,Set(CONNECTEDLINE(name)=${DB(AMPUSER/300/cidname)}${PRESENCESTATE_DISPLAY})
exten => 300,n,Set(FM_DIALSTATUS=${EXTENSION_STATE(300@ext-local)})
exten => 300,n,Set(__EXTTOCALL=${EXTEN})
exten => 300,n,Set(__PICKUPMARK=${EXTEN})
exten => 300,n,Macro(blkvm-setifempty,)
exten => 300,n,GotoIf($["${GOSUB_RETVAL}" = "TRUE"]?skipov)
exten => 300,n,Macro(blkvm-set,reset)
exten => 300,n,Set(__NODEST=)
exten => 300,n(skipov),Set(RRNODEST=${NODEST})
exten => 300,n(skipvmblk),Set(__NODEST=${EXTEN})
exten => 300,n,GosubIf($[${DB_EXISTS(AMPUSER/300/followme/changecid)} = 1 & "${DB(AMPUSER/300/followme/changecid)}" != "default" & "${DB(AMPUSER/300/followme/changecid)}" != ""]?sub-fmsetcid,s,1())
exten => 300,n,Set(RecordMethod=Group)
exten => 300,n(checkrecord),Gosub(sub-record-check,s,1(exten,300,))
exten => 300,n(skipsimple),Set(RingGroupMethod=ringallv2-prim)
exten => 300,n,Set(_FMGRP=300)
exten => 300,n(DIALGRP),GotoIf($[("${DB(AMPUSER/300/followme/grpconf)}"="ENABLED") | ("${FORCE_CONFIRM}"!="") ]?doconfirm)
exten => 300,n,Macro(dial,$[ ${DB(AMPUSER/300/followme/grptime)} + ${DB(AMPUSER/300/followme/prering)} ],${DIAL_OPTIONS},${DB(AMPUSER/300/followme/grplist)})
exten => 300,n,Goto(nextstep)
exten => 300,n(doconfirm),Macro(dial-confirm,$[ ${DB(AMPUSER/300/followme/grptime)} + ${DB(AMPUSER/300/followme/prering)} ],${DIAL_OPTIONS},${DB(AMPUSER/300/followme/grplist)},300)
exten => 300,n(nextstep),Set(RingGroupMethod=)
exten => 300,n,GotoIf($["foo${RRNODEST}" != "foo"]?nodest)
exten => 300,n,Set(__NODEST=)
exten => 300,n,Set(__PICKUPMARK=)
exten => 300,n,Macro(blkvm-clr,)
exten => 300,n,Set(DIALSTATUS=${IF($["${FM_DIALSTATUS}"="NOT_INUSE"&"${DIALSTATUS}"!="CHANUNAVAIL"]?NOANSWER:${IF($["${DIALSTATUS}"="CHANUNAVAIL"|"${FM_DIALSTATUS}"="UNAVAILABLE"|"${FM_DIALSTATUS}"="UNKNOWN"|"${FM_DIALSTATUS}"="INVALID"]?CHANUNAVAIL:BUSY)})})
exten => 300,n,Goto(ext-local,300,dest)
exten => 300,n(nodest),Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST})
;--== end of [ext-findmefollow] ==--;
[fmgrps]
exten => _RG-300.,1,NoCDR()
exten => _RG-300.,n,Macro(dial,${DB(AMPUSER/300/followme/grptime)},${DIAL_OPTIONS}M(confirm^^^300),${EXTEN:7})
;--== end of [fmgrps] ==--;
[sub-fmsetcid]
exten => s,1,Goto(s-${DB(AMPUSER/${EXTTOCALL}/followme/changecid)},1)
exten => s-fixed,1,ExecIf($["${REGEX("^[\+]?[0-9]+$" ${DB(AMPUSER/${EXTTOCALL}/followme/fixedcid)})}" = "1"]?Set(__TRUNKCIDOVERRIDE=${DB(AMPUSER/${EXTTOCALL}/followme/fixedcid)}))
exten => s-fixed,n,Return()
exten => s-extern,1,ExecIf($["${REGEX("^[\+]?[0-9]+$" ${DB(AMPUSER/${EXTTOCALL}/followme/fixedcid)})}" == "1" & "${FROM_DID}" != ""]?Set(__TRUNKCIDOVERRIDE=${DB(AMPUSER/${EXTTOCALL}/followme/fixedcid)}))
exten => s-extern,n,Return()
exten => s-did,1,ExecIf($["${REGEX("^[\+]?[0-9]+$" ${FROM_DID})}" = "1"]?Set(__REALCALLERIDNUM=${FROM_DID}))
exten => s-did,n,Return()
exten => s-forcedid,1,ExecIf($["${REGEX("^[\+]?[0-9]+$" ${FROM_DID})}" = "1"]?Set(__TRUNKCIDOVERRIDE=${FROM_DID}))
exten => s-forcedid,n,Return()
exten => _s-.,1,Noop(Unknown value for AMPUSER/${EXTTOCALL}/followme/changecid of ${DB(AMPUSER/${EXTTOCALL}/followme/changecid)} set to "default")
exten => _s-.,n,Set(DB(AMPUSER/${EXTTOCALL}/followme/changecid)=default)
exten => _s-.,n,Return()
;--== end of [sub-fmsetcid] ==--;
[app-hotelwakeup]
exten => *68,1,Macro(user-callerid,)
exten => *68,n,Answer
exten => *68,n,Wait(1)
exten => *68,n,AGI(wakeupphp)
exten => *68,n,Hangup
;--== end of [app-hotelwakeup] ==--;
[app-calltrace]
exten => *69,1,Goto(app-calltrace-perform,s,1)
;--== end of [app-calltrace] ==--;
[app-calltrace-perform]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Macro(user-callerid,)
exten => s,n,Playback(info-about-last-call&telephone-number)
exten => s,n,Set(lastcaller=${DB(CALLTRACE/${AMPUSER})})
exten => s,n,GotoIf($[ $[ "${lastcaller}" = "" ] | $[ "${lastcaller}" = "unknown" ] ]?noinfo)
exten => s,n,SayDigits(${lastcaller})
exten => s,n,Set(TIMEOUT(digit)=3)
exten => s,n,Set(TIMEOUT(response)=7)
exten => s,n,Background(to-call-this-number&press-1)
exten => s,n,Goto(fin)
exten => s,n(noinfo),Playback(from-unknown-caller)
exten => s,n,Macro(hangupcall,)
exten => s,n(fin),Noop(Waiting for input)
exten => s,n,WaitExten(60,)
exten => s,n,Playback(sorry-youre-having-problems&goodbye)
exten => 1,1,Goto(from-internal,${lastcaller},1)
exten => i,1,Playback(vm-goodbye)
exten => i,n,Macro(hangupcall,)
exten => t,1,Playback(vm-goodbye)
exten => t,n,Macro(hangupcall,)
;--== end of [app-calltrace-perform] ==--;
[app-speakextennum]
exten => *65,1,Answer
exten => *65,n,Wait(1)
exten => *65,n,Macro(user-callerid,)
exten => *65,n,GotoIf($[${DIALPLAN_EXISTS(app-speakextennum,${CHANNEL(language)},1)}]?app-speakextennum,${CHANNEL(language)},1:app-speakextennum,en,1)
exten => en,1,Playback(your)
exten => en,n,Playback(extension)
exten => en,n,Playback(number)
exten => en,n,Playback(is)
exten => en,n,SayDigits(${AMPUSER})
exten => en,n,Wait(2)
exten => en,n,Hangup
exten => ja,1,Playback(your)
exten => ja,n,Playback(extension)
exten => ja,n,Playback(jp-wa)
exten => ja,n,SayDigits(${AMPUSER})
exten => ja,n,Wait(2)
exten => ja,n,Hangup
;--== end of [app-speakextennum] ==--;
[sub-hr12format]
exten => s,1,GotoIf($[${DIALPLAN_EXISTS(sub-hr12format,${CHANNEL(language)},1)}]?sub-hr12format,${CHANNEL(language)},1:sub-hr12format,en,1)
exten => s,n,GotoIf($[${DIALPLAN_EXISTS(sub-hr12format,${CHANNEL(language)},1)}]?${CHANNEL(language)},1:en,1)
exten => en,1,Playback(at-tone-time-exactly)
exten => en,n,SayUnixTime(${FutureTime},,IM 'and' S 'seconds' p)
exten => en,n,Return()
exten => en,n,Playback(at-tone-time-exactly)
exten => en,n,SayUnixTime(${FutureTime},,IM 'and' S 'seconds' p)
exten => en,n,Return()
exten => de,1,Playback(at-tone-time-exactly)
exten => de,n,SayUnixTime(${FutureTime},,IMSp)
exten => de,n,Return()
exten => de,n,Playback(at-tone-time-exactly)
exten => de,n,SayUnixTime(${FutureTime},,IMSp)
exten => de,n,Return()
exten => ja,1,Playback(at-tone-time-exactly)
exten => ja,n,SayUnixTime(${FutureTime},,pIMS)
exten => ja,n,Return()
exten => ja,n,Playback(at-tone-time-exactly)
exten => ja,n,SayUnixTime(${FutureTime},,pIMS)
exten => ja,n,Return()
;--== end of [sub-hr12format] ==--;
[app-speakingclock]
exten => *60,1,Macro(user-callerid,)
exten => *60,n,Answer
exten => *60,n,Wait(1)
exten => *60,n,Set(NumLoops=0)
exten => *60,n(start),Set(FutureTime=$[${EPOCH} + 11])
exten => *60,n,Gosub(sub-hr12format,s,1())
exten => *60,n(waitloop),Set(TimeLeft=$[${FutureTime} - ${EPOCH}])
exten => *60,n,GotoIf($[${TimeLeft} < 1]?playbeep)
exten => *60,n,Wait(1)
exten => *60,n,Goto(waitloop)
exten => *60,n(playbeep),Playback(beep)
exten => *60,n,Wait(5)
exten => *60,n,Set(NumLoops=$[${NumLoops} + 1])
exten => *60,n,GotoIf($[${NumLoops} < 5]?start)
exten => *60,n,Playback(goodbye)
exten => *60,n,Hangup
;--== end of [app-speakingclock] ==--;
[sub-hr24format]
exten => s,1,GotoIf($[${DIALPLAN_EXISTS(sub-hr24format,${CHANNEL(language)},1)}]?${CHANNEL(language)},1:en,1)
exten => en,1,Playback(at-tone-time-exactly)
exten => en,n,SayUnixTime(${FutureTime},,kM 'and' S 'seconds')
exten => en,n,Return()
exten => de,1,Playback(at-tone-time-exactly)
exten => de,n,SayUnixTime(${FutureTime},,kMS)
exten => de,n,Return()
exten => ja,1,Playback(at-tone-time-exactly)
exten => ja,n,SayUnixTime(${FutureTime},,kMS)
exten => ja,n,Return()
;--== end of [sub-hr24format] ==--;
[app-echo-test]
exten => *43,1,Answer
exten => *43,n,Macro(user-callerid,)
exten => *43,n,Wait(1)
exten => *43,n,Background(demo-echotest,,,app-echo-test-echo)
exten => *43,n,Goto(app-echo-test-echo,1,1)
;--== end of [app-echo-test] ==--;
[app-echo-test-echo]
exten => _[0-9#*],1,Echo()
exten => _[0-9#*],n,Playback(demo-echodone)
exten => _[0-9#*],n,Hangup
;--== end of [app-echo-test-echo] ==--;
[macro-outisbusy]
exten => s,1,Progress
exten => s,n,GotoIf($["${EMERGENCYROUTE}" = "YES"]?emergency,1)
exten => s,n,GotoIf($["${INTRACOMPANYROUTE}" = "YES"]?intracompany,1)
exten => s,n,Playback(all-circuits-busy-now&pls-try-call-later, noanswer)
exten => s,n,Congestion(20)
exten => s,n,Hangup
exten => intracompany,1,Playback(all-circuits-busy-now&pls-try-call-later, noanswer)
exten => intracompany,n,Congestion(20)
exten => intracompany,n,Hangup
exten => emergency,1,Playback(all-circuits-busy-now&pls-try-call-later)
exten => emergency,n,Congestion(20)
exten => emergency,n,Hangup
;--== end of [macro-outisbusy] ==--;
[ext-intercom]
exten => _*80.,1,Macro(user-callerid,)
exten => _*80.,n,Set(dialnumber=${EXTEN:3})
exten => _*80.,n,Set(INTERCOM_CALL=TRUE)
exten => _*80.,n,Gosub(sub-record-check,s,1(exten,${dialnumber}))
exten => _*80.,n,GotoIf($["${DB(AMPUSER/${AMPUSER}/intercom/block)}" = "blocked"]?end)
exten => _*80.,n,GotoIf($["${DB(DND/${dialnumber})}" = "YES"]?end)
exten => _*80.,n,GotoIf($["${DB(AMPUSER/${dialnumber}/intercom/${AMPUSER})}" = "allow" ]?allow)
exten => _*80.,n,GotoIf($["${DB(AMPUSER/${dialnumber}/intercom/${AMPUSER})}" = "deny" ]?nointercom)
exten => _*80.,n,GotoIf($["${DB(AMPUSER/${dialnumber}/intercom)}" = "disabled" ]?nointercom)
exten => _*80.,n(allow),Set(DEVICES=${DB(AMPUSER/${dialnumber}/device)})
exten => _*80.,n,GotoIf($["${DEVICES}" = "" ]?end)
exten => _*80.,n,Set(OVERRIDE=${DB(AMPUSER/${dialnumber}/intercom/override)})
exten => _*80.,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => _*80.,n,Set(_SIPURI=)
exten => _*80.,n,Set(_ALERTINFO=Ring Answer)
exten => _*80.,n,Set(_CALLINFO=<uri>\;answer-after=0)
exten => _*80.,n,Set(_SIPURI=intercom=true)
exten => _*80.,n,Set(_DTIME=5)
exten => _*80.,n,Set(_ANSWERMACRO=)
exten => _*80.,n,GotoIf($[${LOOPCNT} > 1 ]?pagemode)
exten => _*80.,n,Macro(autoanswer,${DEVICES})
exten => _*80.,n,Set(_DOPTIONS=A(beep)b(autoanswer^s^1(${ALERTINFO},${CALLINFO})))
exten => _*80.,n(check),ChanIsAvail(${DEVICE},s)
exten => _*80.,n,ExecIf($["${AVAILSTATUS}" = "6"]?Set(AVAILORIGCHAN=${DEVICE}))
exten => _*80.,n,GotoIf($["${AVAILORIGCHAN}" != ""]?continue)
exten => _*80.,n,ExecIf($["${OVERRIDE}" = ""]?Set(OVERRIDE=reject))
exten => _*80.,n,GotoIf($["${OVERRIDE}" = "reject"]?end)
exten => _*80.,n,Set(DTIME=30)
exten => _*80.,n,ExecIf($["${OVERRIDE}" = "ring"]?Set(DOPTIONS=A(beep)))
exten => _*80.,n(continue),Noop(Continuing with page)
exten => _*80.,n,GotoIf($["${DB(AMPUSER/${EXTEN:3}/cidname)}" = ""]?godial)
exten => _*80.,n,Set(CONNECTEDLINE(name,i)=${DB(AMPUSER/${EXTEN:3}/cidname)})
exten => _*80.,n,Set(CONNECTEDLINE(num)=${EXTEN:3})
exten => _*80.,n(godial),Dial(${DIAL},${DTIME},I${DOPTIONS}${INTERCOM_EXT_DOPTIONS})
exten => _*80.,n(end),ExecIf($[${INTERCOM_RETURN}]?Return())
exten => _*80.,n,Busy(20)
exten => _*80.,n,Macro(hangupcall,)
exten => _*80.,n(pagemode),Set(ITER=1)
exten => _*80.,n,Set(DIALSTR=)
exten => _*80.,n(begin),ChanIsAvail(${DB(DEVICE/${CUT(DEVICES,&,${ITER})}/dial)},s)
exten => _*80.,n,GotoIf($["${AVAILORIGCHAN}" = ""]?skip)
exten => _*80.,n,Set(DIALSTR=${DIALSTR}-${CUT(DEVICES,&,${ITER})})
exten => _*80.,n(skip),Set(ITER=$[${ITER} + 1])
exten => _*80.,n,GotoIf($[${ITER} <= ${LOOPCNT}]?begin)
exten => _*80.,n,Set(DIALSTR=${DIALSTR:1})
exten => _*80.,n,GotoIf($["${DIALSTR}" = ""]?end2)
exten => _*80.,n,Set(_AMPUSER=${AMPUSER})
exten => _*80.,n,Gosub(page,1(${DIALSTR}))
exten => _*80.,n(end2),ExecIf($[${INTERCOM_RETURN}]?Return())
exten => _*80.,n,Busy(20)
exten => _*80.,n,Macro(hangupcall,)
exten => _*80.,n(nointercom),Noop(Intercom disallowed by ${dialnumber})
exten => _*80.,n,ExecIf($[${INTERCOM_RETURN}]?Return())
exten => _*80.,n,Playback(intercom&for&extension)
exten => _*80.,n,SayDigits(${dialnumber})
exten => _*80.,n,Playback(is&disabled)
exten => _*80.,n,Congestion(20)
exten => page,1,Set(PAGE_CONF=${EPOCH}${RAND(100,999)})
exten => page,n,Set(PAGEMODE=PAGE)
exten => page,n,Set(PAGE_MEMBERS=${ARG1})
exten => page,n,Set(PAGE_CONF_OPTS=duplex)
exten => page,n,AGI(page.agi)
exten => page,n,Set(CONFBRIDGE(user,template)=page_user_duplex)
exten => page,n,Set(CONFBRIDGE(user,admin)=yes)
exten => page,n,Set(CONFBRIDGE(user,marked)=yes)
exten => page,n,ConfBridge(${PAGE_CONF},,,admin_menu)
exten => page,n,Hangup
;--== end of [ext-intercom] ==--;
[ext-intercom-users]
exten => *80300,1,Goto(ext-intercom,${EXTEN},1)
exten => *54,1,Answer
exten => *54,n,Wait(1)
exten => *54,n,Macro(user-callerid,)
exten => *54,n,Set(DB(AMPUSER/${AMPUSER}/intercom)=enabled)
exten => *54,n,Playback(intercom&enabled)
exten => *54,n,Macro(hangupcall,)
exten => _*54.,1,Answer
exten => _*54.,n,Wait(1)
exten => _*54.,n,Macro(user-callerid,)
exten => _*54.,n,GotoIf($["${DB(AMPUSER/${AMPUSER}/intercom/${EXTEN:3})}" = "allow" ]}?unset)
exten => _*54.,n,GotoIf($[${DB_EXISTS(AMPUSER/${EXTEN:3}/device)} != 1]?invaliduser)
exten => _*54.,n,Set(DB(AMPUSER/${AMPUSER}/intercom/${EXTEN:3})=allow)
exten => _*54.,n,Playback(intercom&enabled&for&extension&number)
exten => _*54.,n,SayDigits(${EXTEN:3})
exten => _*54.,n,Macro(hangupcall,)
exten => _*54.,n(unset),dbDeltree(AMPUSER/${AMPUSER}/intercom/${EXTEN:3})
exten => _*54.,n,Playback(intercom&enabled&cancelled&for&extension&number)
exten => _*54.,n,SayDigits(${EXTEN:3})
exten => _*54.,n,Macro(hangupcall,)
exten => _*54.,n(invaliduser),Playback(extension&number)
exten => _*54.,n,SayDigits(${EXTEN:3})
exten => _*54.,n,Playback(is&invalid)
exten => _*54.,n,Macro(hangupcall,)
exten => *55,1,Answer
exten => *55,n,Wait(1)
exten => *55,n,Macro(user-callerid,)
exten => *55,n,Set(DB(AMPUSER/${AMPUSER}/intercom)=disabled)
exten => *55,n,Playback(intercom&disabled)
exten => *55,n,Macro(hangupcall,)
exten => _*55.,1,Answer
exten => _*55.,n,Wait(1)
exten => _*55.,n,Macro(user-callerid,)
exten => _*55.,n,GotoIf($["${DB(AMPUSER/${AMPUSER}/intercom/${EXTEN:3})}" = "deny" ]}?unset2)
exten => _*55.,n,GotoIf($[${DB_EXISTS(AMPUSER/${EXTEN:3}/device)} != 1]?invaliduser2)
exten => _*55.,n,Set(DB(AMPUSER/${AMPUSER}/intercom/${EXTEN:3})=deny)
exten => _*55.,n,Playback(intercom&disabled&for&extension&number)
exten => _*55.,n,SayDigits(${EXTEN:3})
exten => _*55.,n,Macro(hangupcall,)
exten => _*55.,n(unset2),dbDeltree(AMPUSER/${AMPUSER}/intercom/${EXTEN:3})
exten => _*55.,n,Playback(intercom&disabled&cancelled&for&extension&number)
exten => _*55.,n,SayDigits(${EXTEN:3})
exten => _*55.,n,Macro(hangupcall,)
exten => _*55.,n(invaliduser2),Playback(extension&number)
exten => _*55.,n,SayDigits(${EXTEN:3})
exten => _*55.,n,Playback(is&invalid)
exten => _*55.,n,Macro(hangupcall,)
;--== end of [ext-intercom-users] ==--;
[macro-autoanswer]
exten => s,1,GotoIf($["${KNOWNDIAL}" != ""]?knowndial)
exten => s,n,Set(DEVICE=${DB(DEVICE/${ARG1}/dial)})
exten => s,n,GotoIf($["${DEVICE:0:5}" == "PJSIP" ]?dopjsip)
exten => s,n,Set(KNOWNDIAL=${DEVICE})
exten => s,n,Goto(knowndial)
exten => s,n(dopjsip),Set(KNOWNDIAL=${PJSIP_DIAL_CONTACTS(${ARG1})})
exten => s,n,GotoIf($[${REGEX("&" ${KNOWNDIAL})} == 0]?knowndial)
exten => s,n,Gosub(app-paging,ssetup,1())
exten => s,n,Set(PAGEMODE=PAGE)
exten => s,n,Set(PAGE_CONF_OPTS=duplex)
exten => s,n,Set(STREAM=NONE)
exten => s,n,Set(PAGE_MEMBERS=${ARG1})
exten => s,n,AGI(page.agi)
exten => s,n,Set(CONFBRIDGE(user,template)=page_user_duplex)
exten => s,n,Set(CONFBRIDGE(user,admin)=yes)
exten => s,n,Set(CONFBRIDGE(user,marked)=yes)
exten => s,n,ConfBridge(${PAGE_CONF},,,admin_menu)
exten => s,n,Hangup
exten => s,n(knowndial),Set(DIAL=${KNOWNDIAL})
exten => s,n,ExecIf($["${DIAL:0:3}" = "ZAP"]?Set(DIAL=DAHDI${DIAL:3}))
exten => s,n,GotoIf($["${DB(DEVICE/${ARG1}/autoanswer/macro)}" != "" ]?macro)
exten => s,n,Set(USERAGENT=${SIPPEER(${CUT(DIAL,/,2)},useragent)})
exten => s,n,ExecIf($["${KNOWNAGENT}" != ""]?Set(USERAGENT=${KNOWNAGENT}))
exten => s,n,ExecIf($["${USERAGENT:0:5}" = "Mitel"]?Set(CALLINFO=<sip:broadworks.net>\;answer-after=0))
exten => s,n,ExecIf($["${USERAGENT:0:9}" = "Panasonic"]?Set(ALERTINFO=Intercom))
exten => s,n,ExecIf($["${USERAGENT:0:7}" = "Polycom"]?Set(ALERTINFO=info=Auto Answer))
exten => s,n,ExecIf($["${USERAGENT:0:6}" = "Digium"]?Set(ALERTINFO=ring-answer))
exten => s,n,ExecIf($["${SIPURI}" != ""]?Set(__SIP_URI_OPTIONS=${SIPURI}))
exten => s,n+2(macro),Macro(${DB(DEVICE/${ARG1}/autoanswer/macro)},${ARG1})
;--== end of [macro-autoanswer] ==--;
[autoanswer]
exten => s,1,GosubIf($["${ARG1}" != ""]?addheader,1(Alert-Info,${ARG1}))
exten => s,n,GosubIf($["${ARG2}" != ""]?addheader,1(Call-Info,${ARG2}))
exten => s,n,Return()
exten => addheader,1,SIPAddHeader(${ARG1}: ${ARG2})
exten => addheader,n,Set(PJSIP_HEADER(add,${ARG1})=${ARG2})
exten => addheader,n,Return()
;--== end of [autoanswer] ==--;
[app-paging]
exten => ssetup,1,Set(_SIPURI=)
exten => ssetup,n,Set(_ALERTINFO=Ring Answer)
exten => ssetup,n,Set(_CALLINFO=<uri>\;answer-after=0)
exten => ssetup,n,Set(_SIPURI=intercom=true)
exten => ssetup,n,Set(_DTIME=5)
exten => ssetup,n,Set(_ANSWERMACRO=)
exten => ssetup,n,Set(PAGE_CONF=${EPOCH}${RAND(100,999)})
exten => ssetup,n,Return()
exten => _PAGE.,1(SKIPCHECK),Macro(autoanswer,${EXTEN:4})
exten => _PAGE.,n,Set(_DOPTIONS=A(beep)b(autoanswer^s^1(${ALERTINFO},${CALLINFO})))
exten => _PAGE.,n,Dial(${DIAL},${DTIME},${DOPTIONS})
exten => _PAGE.,n(skipself),Hangup
exten => _SPAGE.,1(chanspy),ChanSpy(${SP_DEVICE}-,qW)
exten => _SPAGE.,n,Hangup
;--== end of [app-paging] ==--;
[app-page-stream]
exten => s,1,Wait(1)
exten => s,n,Answer
exten => s,n,Set(CONFBRIDGE(user,template)=page_user_duplex)
exten => s,n,Set(CONFBRIDGE(user,marked)=yes)
exten => s,n,ConfBridge(${PAGE_CONF},,,)
exten => s,n,Hangup
;--== end of [app-page-stream] ==--;
[macro-parked-call]
exten => s,1,Macro(user-callerid,)
exten => s,n,Gosub(sub-record-check,s,1(parking,${AMPUSER},${AMPUSER}))
exten => s,n,Noop(PARKRETURNTO: ${SHARED(PARKRETURNTO,${CHANNEL})})
exten => s,n,GotoIf($[${LEN(${SHARED(PARKRETURNTO,${CHANNEL})})} > 0]?backtosender)
exten => s,n,GotoIf($[${LEN(${BLINDTRANSFER})} > 0 | ${LEN(${ATTENDEDTRANSFER})} > 0]?attemptpark)
exten => s,n,AGI(parkfetch.agi,${ARG1},${ARG2})
exten => s,n,GotoIf($["${REC_STATUS}" != "RECORDING"]?next)
exten => s,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => s,n,Set(CDR(recordingfile)=${CALLFILENAME}.${MON_FMT})
exten => s,n(next),Set(CCSS_SETUP=TRUE)
exten => s,n,GotoIf($["${ARG1}" = "" | ${DIALPLAN_EXISTS(${IF($["${ARG2}" = "default"]?parkedcalls:${ARG2})},${ARG1},1)} = 1]?pcall)
exten => s,n,ResetCDR()
exten => s,n,NoCDR()
exten => s,n,Wait(1)
exten => s,n,Playback(pbx-invalidpark)
exten => s,n,Wait(1)
exten => s,n,Hangup
exten => s,n(pcall),Noop(User: ${CALLERID(all)} attempting to pick up Parked Call Slot ${ARG1})
exten => s,n,Noop(PARKIE: ${PARKIE})
exten => s,n,Set(SHARED(PARKRETURNTO,${PARKIE})=)
exten => s,n,Set(PARKOWNER=1)
exten => s,n,ParkedCall(${ARG2},${ARG1})
exten => s,n(attemptpark),Noop(User: ${CALLERID(all)} attempting to Park into slot ${ARG1})
exten => s,n,Noop(Blind Transfer: ${BLINDTRANSFER}, Attended Transfer: ${ATTENDEDTRANSFER})
exten => s,n,Noop($[${LEN(${PARKOWNER})} = 0])
exten => s,n,GotoIf($[${LEN(${PARKOWNER})} = 0]?parkit)
exten => s,n,Macro(hangupcall,)
exten => s,n(parkit),Set(PARKINGEXTEN=${ARG1})
exten => s,n,ExecIf($[${LEN(${BLINDTRANSFER})} > 0]?Set(SHARED(PARKRETURNTO,${CHANNEL})=${CUT(BLINDTRANSFER,-,1)}):Set(SHARED(PARKRETURNTO,${CHANNEL})=${CUT(ATTENDEDTRANSFER,-,1)}))
exten => s,n,Noop(PARKRETURNTO: ${SHARED(PARKRETURNTO,${CHANNEL})})
exten => s,n,Park(${ARG2},sc(${CONTEXT},s,200))
exten => s,1+199(backtosender),Noop(Attempting to go back to sender)
exten => s,n,Set(PARKCALLBACK=${PARKER})
exten => s,n,Set(SHARED(PARKRETURNTO,${CHANNEL})=)
exten => s,n,Goto(park-return-routing,${PARKINGSLOT},1)
exten => h,1,Macro(hangupcall,)
;--== end of [macro-parked-call] ==--;
[parkedcallstimeout]
exten => _[0-9a-zA-Z*#].,1,Set(PARKCALLBACK=${REPLACE(EXTEN,_,/)})
exten => _[0-9a-zA-Z*#].,n,GotoIf($["${REC_STATUS}" != "RECORDING"]?next)
exten => _[0-9a-zA-Z*#].,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => _[0-9a-zA-Z*#].,n,MixMonitor(${MIXMON_DIR}${YEAR}/${MONTH}/${DAY}/${CALLFILENAME}.${MIXMON_FORMAT},a,${MIXMON_POST})
exten => _[0-9a-zA-Z*#].,n(next),Goto(park-return-routing,${PARKINGSLOT},1)
;--== end of [parkedcallstimeout] ==--;
[park-dial]
exten => t,1,Noop(WARNING: PARKRETURN to: [${EXTEN}] failed with: [${DIALSTATUS}]. Trying Alternate Dest On Parking Lot ${PARKINGSLOT})
exten => t,n,Goto(park-orphan-routing,70,1)
exten => _[0-9a-zA-Z*#].,1,Noop(WARNING: PARKRETURN to: [${EXTEN}] failed with: [${DIALSTATUS}]. Trying Alternate Dest On Parking Lot ${PARKINGSLOT})
exten => _[0-9a-zA-Z*#].,n,Goto(park-orphan-routing,70,1)
;--== end of [park-dial] ==--;
[park-orphan-routing]
exten => 70,1,Goto(app-blackhole,hangup,1)
;--== end of [park-orphan-routing] ==--;
[park-return-routing]
exten => 70,1,Set(PLOT=70)
exten => 70,n,Dial(${PARKCALLBACK},15,tr)
exten => 70,n,Set(PARKCALLBACK=)
exten => 70,n,Goto(app-blackhole,hangup,1)
exten => 71,1,Goto(70,1)
exten => 72,1,Goto(70,1)
exten => 73,1,Goto(70,1)
exten => 74,1,Goto(70,1)
exten => 75,1,Goto(70,1)
exten => 76,1,Goto(70,1)
exten => 77,1,Goto(70,1)
exten => 78,1,Goto(70,1)
;--== end of [park-return-routing] ==--;
[park-hints]
include => parkedcalls ; Default Lot
exten => 71,1,Macro(parked-call,71,default)
exten => 71,hint,park:71@parkedcalls
exten => 72,1,Macro(parked-call,72,default)
exten => 72,hint,park:72@parkedcalls
exten => 73,1,Macro(parked-call,73,default)
exten => 73,hint,park:73@parkedcalls
exten => 74,1,Macro(parked-call,74,default)
exten => 74,hint,park:74@parkedcalls
exten => 75,1,Macro(parked-call,75,default)
exten => 75,hint,park:75@parkedcalls
exten => 76,1,Macro(parked-call,76,default)
exten => 76,hint,park:76@parkedcalls
exten => 77,1,Macro(parked-call,77,default)
exten => 77,hint,park:77@parkedcalls
exten => 78,1,Macro(parked-call,78,default)
exten => 78,hint,park:78@parkedcalls
exten => *85,1,Macro(parked-call,,default)
exten => *85,hint,park:71@parkedcalls&park:72@parkedcalls&park:73@parkedcalls&park:74@parkedcalls&park:75@parkedcalls&park:76@parkedcalls&park:77@parkedcalls&park:78@parkedcalls
exten => *8570,1,Macro(parked-call,,default)
exten => *8570,hint,park:71@parkedcalls&park:72@parkedcalls&park:73@parkedcalls&park:74@parkedcalls&park:75@parkedcalls&park:76@parkedcalls&park:77@parkedcalls&park:78@parkedcalls
;--== end of [park-hints] ==--;
[app-pbdirectory]
exten => 411,1,Goto(pbdirectory,1)
exten => pbdirectory,1,Answer
exten => pbdirectory,n,Wait(1)
exten => pbdirectory,n,Macro(user-callerid,)
exten => pbdirectory,n,AGI(pbdirectory)
exten => pbdirectory,n,GotoIf($["${dialnumber}"=""]?hangup,1)
exten => pbdirectory,n,Noop(Got number to dial: ${dialnumber})
exten => pbdirectory,n,Dial(Local/${dialnumber}@from-internal/n,)
exten => hangup,1,Hangup
;--== end of [app-pbdirectory] ==--;
[app-recordings]
exten => *77,1,Macro(user-callerid,)
exten => *77,n,Wait(2)
exten => *77,n,Macro(systemrecording,dorecord)
exten => *99,1,Macro(user-callerid,)
exten => *99,n,Wait(2)
exten => *99,n,Macro(systemrecording,docheck)
;--== end of [app-recordings] ==--;
[macro-systemrecording]
exten => s,1,Set(RECFILE=${IF($["${ARG2}" = ""]?/var/spool/asterisk/tmp/${AMPUSER}-ivrrecording:${ARG2})})
exten => s,n,ExecIf($["${ARG3}" != ""]?Authenticate(${ARG3}))
exten => s,n,Goto(${ARG1},1)
exten => dorecord,1,GotoIf($["${ARG2}" = ""]?skipremove)
exten => dorecord,n,System(rm ${ASTVARLIBDIR}/sounds/${RECFILE}.*)
exten => dorecord,n(skipremove),Record(${RECFILE}.wav,,,k)
exten => dorecord,n,Wait(1)
exten => dorecord,n,Goto(confmenu,1)
exten => docheck,1,Playback(beep)
exten => docheck,n(dc_start),Background(${RECFILE},m,${CHANNEL(language)},macro-systemrecording)
exten => docheck,n,Wait(1)
exten => docheck,n,Goto(confmenu,1)
exten => confmenu,1,Background(to-listen-to-it&press-1&to-rerecord-it&press-star&astcc-followed-by-pound,m,${CHANNEL(language)},macro-systemrecording)
exten => confmenu,n,Read(RECRESULT,,1,,,4)
exten => confmenu,n,GotoIf($["x${RECRESULT}"="x*"]?dorecord,1)
exten => confmenu,n,GotoIf($["x${RECRESULT}"="x1"]?docheck,2)
exten => confmenu,n,Goto(1)
exten => 1,1,Goto(docheck,dc_start)
exten => *,1,Goto(dorecord,1)
exten => t,1,Playback(goodbye)
exten => t,n,Hangup
exten => i,1,Playback(pm-invalid-option)
exten => i,n,Goto(confmenu,1)
exten => h,1,Hangup
;--== end of [macro-systemrecording] ==--;
[ext-group]
exten => h,1,Macro(hangupcall,)
;--== end of [ext-group] ==--;
[macro-speeddial-lookup]
exten => s,1,GotoIf($["${ARG2}"=""]]?lookupsys)
exten => s,n,Set(SPEEDDIALNUMBER=)
exten => s,n(lookupuser),Set(SPEEDDIALNUMBER=${DB(AMPUSER/${ARG2}/speeddials/${ARG1})})
exten => s,n,GotoIf($["${SPEEDDIALNUMBER}"=""]?lookupsys)
exten => s,n,Noop(Found speeddial ${ARG1} for user ${ARG2}: ${SPEEDDIALNUMBER})
exten => s,n,Goto(end)
exten => s,lookupuser+101(lookupsys),Set(SPEEDDIALNUMBER=${DB(sysspeeddials/${ARG1})})
exten => s,n,GotoIf($["${SPEEDDIALNUMBER}"=""]?failed)
exten => s,n,Noop(Found system speeddial ${ARG1}: ${SPEEDDIALNUMBER})
exten => s,n,Goto(end)
exten => s,lookupsys+101(failed),Noop(No system or user speeddial found)
exten => s,n(end),Noop(End of Speeddial-lookup)
;--== end of [macro-speeddial-lookup] ==--;
[app-speeddial]
exten => _*0.,1,Macro(user-callerid,)
exten => _*0.,n,Set(SPEEDDIALLOCATION=${EXTEN:2})
exten => _*0.,n(lookup),Macro(speeddial-lookup,${SPEEDDIALLOCATION},${AMPUSER})
exten => _*0.,n,GotoIf($["${SPEEDDIALNUMBER}"=""]?failed)
exten => _*0.,n,Goto(from-internal,${SPEEDDIALNUMBER},1)
exten => _*0.,lookup+101(failed),Playback(speed-dial-empty)
exten => _*0.,n,Congestion()
exten => *75,1,Goto(app-speeddial-set,s,1)
;--== end of [app-speeddial] ==--;
[app-speeddial-set]
exten => lang-playback,1,GosubIf($[${DIALPLAN_EXISTS(app-speeddial-set,${CHANNEL(language)})}]?app-speeddial-set,${CHANNEL(language)},${ARG1}():app-speeddial-set,en,${ARG1}())
exten => lang-playback,n,Return()
exten => s,1,Macro(user-callerid,)
exten => s,n(setloc),Read(newlocation,speed-enterlocation,,,,)
exten => s,n(lookup),Macro(speeddial-lookup,${newlocation},${AMPUSER})
exten => s,n,GotoIf($["${SPEEDDIALNUMBER}"!=""]?conflicts)
exten => s,n(setnum),Read(newnum,speed-enternumber,,,,)
exten => s,n(success),Set(DB(AMPUSER/${AMPUSER}/speeddials/${newlocation})=${newnum})
exten => s,n,Gosub(app-speeddial-set,lang-playback,1(hook_0))
exten => s,n,Hangup
exten => s,n(conflicts),Gosub(app-speeddial-set,lang-playback,1(hook_1))
exten => s,n,WaitExten(60,)
exten => 1,1,Gosub(app-speeddial-set,lang-playback,1(hook_2))
exten => 1,n,Goto(s,conflicts)
exten => 2,1,Goto(s,setloc)
exten => 3,1,Goto(s,setnum)
exten => t,1,Congestion()
exten => en,1(hook_0),Playback(speed-dial)
exten => en,n,SayDigits(${newlocation})
exten => en,n,Playback(is-set-to)
exten => en,n,SayDigits(${newnum})
exten => en,n,Return()
exten => en,n(hook_1),Playback(speed-dial)
exten => en,n,SayDigits(${newlocation})
exten => en,n,Playback(is-in-use)
exten => en,n,Background(press-1&to-listen-to-it&press-2&to-enter-a-diff&location&press-3&to-change&telephone-number)
exten => en,n,Return()
exten => en,n(hook_2),Playback(speed-dial)
exten => en,n,SayDigits(${newlocation})
exten => en,n,Playback(is-set-to)
exten => en,n,SayDigits(${SPEEDDIALNUMBER})
exten => en,n,Return()
exten => ja,1(hook_0),Playback(speed-dial)
exten => ja,n,SayDigits(${newlocation})
exten => ja,n,Playback(jp-wo)
exten => ja,n,SayDigits(${newnum})
exten => ja,n,Playback(is-set-to)
exten => ja,n,Return()
exten => ja,n(hook_1),Playback(speed-dial)
exten => ja,n,SayDigits(${newlocation})
exten => ja,n,Playback(jp-wa&is-in-use)
exten => ja,n,Background(list&press-1&to-enter-a-diff&location&jp-wo&to-enter&press-2&telephone-number&jp-wo&to-change&press-3)
exten => ja,n,Return()
exten => ja,n(hook_2),Playback(speed-dial)
exten => ja,n,SayDigits(${newlocation})
exten => ja,n,Playback(jp-wa)
exten => ja,n,SayDigits(${SPEEDDIALNUMBER})
exten => ja,n,Playback(is-set-to-2)
exten => ja,n,Return()
;--== end of [app-speeddial-set] ==--;
[vm-callme]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n(repeat),Background(${MSG}&silence/2&vm-repeat&vm-starmain)
exten => s,n,WaitExten(15,)
exten => 5,1,Goto(s,repeat)
exten => #,1,Playback(vm-goodbye)
exten => #,n,Hangup
exten => *,1,Macro(get-vmcontext,${MBOX})
exten => *,n,VoiceMailMain(${MBOX}@${VMCONTEXT},s)
exten => i,1,Playback(pm-invalid-option)
exten => i,n,Goto(s,repeat)
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup
exten => h,1,Hangup
;--== end of [vm-callme] ==--;
[app-dialvm]
exten => *98,1,Macro(user-callerid,)
exten => *98,n,Answer
exten => *98,n(start),Wait(1)
exten => *98,n,Noop(app-dialvm: Asking for mailbox)
exten => *98,n,Read(MAILBOX,vm-login,,,3,2)
exten => *98,n(check),Noop(app-dialvm: Got Mailbox ${MAILBOX})
exten => *98,n,Macro(get-vmcontext,${MAILBOX})
exten => *98,n,Set(VMBOXEXISTSSTATUS=${IF(${VM_INFO(${MAILBOX}@${VMCONTEXT},exists)}?SUCCESS:FAILED)})
exten => *98,n,GotoIf($["${VMBOXEXISTSSTATUS}" = "SUCCESS"]?good:bad)
exten => *98,n,Macro(hangupcall,)
exten => *98,n(good),Noop(app-dialvm: Good mailbox ${MAILBOX}@${VMCONTEXT})
exten => *98,n,VoiceMailMain(${MAILBOX}@${VMCONTEXT})
exten => *98,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => *98,n,Macro(hangupcall,)
exten => *98,n(bad),Noop(app-dialvm: BAD mailbox ${MAILBOX}@${VMCONTEXT})
exten => *98,n,Wait(1)
exten => *98,n,Noop(app-dialvm: Asking for password so people can't probe for existence of a mailbox)
exten => *98,n,Read(FAKEPW,vm-password,,,3,2)
exten => *98,n,Noop(app-dialvm: Asking for mailbox again)
exten => *98,n,Read(MAILBOX,vm-incorrect-mailbox,,,3,2)
exten => *98,n,Goto(check)
exten => *98,n,Macro(hangupcall,)
exten => *98,n(playret),Playback(beep&you-will-be-transfered-menu&silence/1)
exten => *98,n,Goto(${IVR_CONTEXT},return,1)
exten => _*98.,1,Answer
exten => _*98.,n,Wait(1)
exten => _*98.,n,Macro(get-vmcontext,${EXTEN:3})
exten => _*98.,n,VoiceMailMain(${EXTEN:3}@${VMCONTEXT})
exten => _*98.,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?${IVR_CONTEXT},return,1)
exten => _*98.,n,Macro(hangupcall,)
;--== end of [app-dialvm] ==--;
[app-vmmain]
exten => *97,1,Answer
exten => *97,n,Wait(1)
exten => *97,n,Macro(user-callerid,)
exten => *97,n,Macro(get-vmcontext,${AMPUSER})
exten => *97,n(check),Set(VMBOXEXISTSSTATUS=${IF(${VM_INFO(${AMPUSER}@${VMCONTEXT},exists)}?SUCCESS:FAILED)})
exten => *97,n,GotoIf($["${VMBOXEXISTSSTATUS}" = "SUCCESS"]?mbexist)
exten => *97,n,VoiceMailMain()
exten => *97,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => *97,n,Macro(hangupcall,)
exten => *97,check+101(mbexist),GotoIf($["${DB(AMPUSER/${AMPUSER}/novmpw)}"!=""]?novmpw:vmpw)
exten => *97,n(novmpw),Noop(Verifying channel ${CHANNEL} is actually ${AMPUSER})
exten => *97,n,GotoIf($["${REGEX("^${DB(DEVICE/${AMPUSER}/dial)}-[0-9a-fx]+$" ${CHANNEL})}"!="1"]?vmpws)
exten => *97,n,VoiceMailMain(${AMPUSER}@${VMCONTEXT},s)
exten => *97,n,Goto(vmend)
exten => *97,n(vmpws),Noop(Channel ${CHANNEL} is NOT ${AMPUSER} forcing VM Password)
exten => *97,n(vmpw),VoiceMailMain(${AMPUSER}@${VMCONTEXT})
exten => *97,n,Goto(vmend)
exten => *97,n(vmend),GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => *97,n,Macro(hangupcall,)
exten => *97,n(playret),Playback(beep&you-will-be-transfered-menu&silence/1)
exten => *97,n,Goto(${IVR_CONTEXT},return,1)
;--== end of [app-vmmain] ==--;
[from-did-direct-ivr]
exten => *300,1,Macro(blkvm-clr,)
exten => *300,n,Set(__NODEST=)
exten => *300,n,Macro(vm,300,DIRECTDIAL,${IVR_RETVM})
exten => *300,n,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?ext-local,vmret,playret)
exten => 300,1,Macro(blkvm-clr,)
exten => 300,n,Set(__NODEST=)
exten => 300,n,Goto(from-did-direct,300,1)
;--== end of [from-did-direct-ivr] ==--;
[app-userlogonoff]
exten => *12,1,Macro(user-logoff,)
exten => *12,n(hook_off),Hangup
exten => *11,1,Macro(user-logon,)
exten => *11,n(hook_on_1),Hangup
exten => _*11.,1,Macro(user-logon,${EXTEN:3},)
exten => _*11.,n(hook_on_2),Hangup
;--== end of [app-userlogonoff] ==--;
[ext-local-confirm]
exten => _LC-.,1,ExecIf($["${DIRECTION}" = "INBOUND"]?Set(DIAL_OPTIONS=${REPLACE(DIAL_OPTIONS,T)}I))
exten => _LC-.,n,ExecIf($["${ALERT_INFO}"!=""]?SIPAddHeader(Alert-Info: ${ALERT_INFO}))
exten => _LC-.,n,Dial(${DB(DEVICE/${EXTEN:3}/dial)},${RT},${DIAL_OPTIONS}M(auto-confirm^${RG_IDX}))
;--== end of [ext-local-confirm] ==--;
[findmefollow-ringallv2]
exten => _FMPR-.,1,NoCDR()
exten => _FMPR-.,n,Set(RingGroupMethod=)
exten => _FMPR-.,n,Set(USE_CONFIRMATION=)
exten => _FMPR-.,n,Set(RINGGROUP_INDEX=)
exten => _FMPR-.,n,Macro(simple-dial,${EXTEN:5},${FMREALPRERING})
exten => _FMPR-.,n,ExecIf($["${DIALSTATUS}" = "BUSY"]?Set(SHARED(FM_DND,${FMUNIQUE})=DND))
exten => _FMPR-.,n,Hangup
exten => _FMGL-.,1,NoCDR()
exten => _FMGL-.,n,Set(ENDLOOP=$[${EPOCH} + ${FMPRERING} + 2])
exten => _FMGL-.,n(start),GotoIf($["${SHARED(FM_DND,${FMUNIQUE})}" = "DND"]?dodnd)
exten => _FMGL-.,n,Wait(1)
exten => _FMGL-.,n,GotoIf($[${EPOCH} < ${ENDLOOP}]?start)
exten => _FMGL-.,n,Set(SHARED(FM_DND,${FMUNIQUE})=)
exten => _FMGL-.,n(dodial),Macro(dial,${FMGRPTIME},${DIAL_OPTIONS},${EXTEN:5})
exten => _FMGL-.,n,Hangup
exten => _FMGL-.,n+10(dodnd),Set(SHARED(FM_DND,${FMUNIQUE})=)
exten => _FMGL-.,n,GotoIf($["${FMPRIME}" = "FALSE"]?dodial)
exten => _FMGL-.,n,Hangup
;--== end of [findmefollow-ringallv2] ==--;
[app-pickup]
exten => _**.,1,Macro(user-callerid,)
exten => _**.,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => _**.,n,Pickup(${EXTEN:2}&${EXTEN:2}@PICKUPMARK)
exten => _**.,n,Hangup
exten => _***80.,1,Macro(user-callerid,)
exten => _***80.,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => _***80.,n,Pickup(${EXTEN:5}&${EXTEN:5}@PICKUPMARK)
exten => _***80.,n,Hangup
;--== end of [app-pickup] ==--;
[app-chanspy]
exten => 555,1,Macro(user-callerid,)
exten => 555,n,Answer
exten => 555,n,Wait(1)
exten => 555,n,ChanSpy()
exten => 555,n,Hangup
;--== end of [app-chanspy] ==--;
[ext-test]
exten => 7777,1,Macro(user-callerid,)
exten => 7777,n,Goto(from-pstn,${EXTEN},1)
exten => h,1,Macro(hangupcall,)
;--== end of [ext-test] ==--;
[ext-did]
include => ext-did-0001
include => ext-did-0002
exten => foo,1,Noop(bar)
;--== end of [ext-did] ==--;
[ext-did-catchall]
exten => s,1,Noop(No DID or CID Match)
exten => s,n(a2),Answer
exten => s,n,Log(WARNING,Friendly Scanner from ${CUT(CUT(SIP_HEADER(Via), ,2),:,1)})
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,SayAlpha(${FROM_DID})
exten => s,n,Hangup
exten => _.,1,Set(__FROM_DID=${EXTEN})
exten => _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
exten => _.,n,Goto(s,a2)
exten => h,1,Hangup
;--== end of [ext-did-catchall] ==--;
[from-trunk-pjsip-Telekom]
exten => _.,1,Set(GROUP()=OUT_2)
exten => _.,n,Goto(from-trunk,${EXTEN},1)
;--== end of [from-trunk-pjsip-Telekom] ==--;
[ext-trunk]
exten => 2,1,Set(TDIAL_SUFFIX=@Telekom)
exten => 2,n,Set(TDIAL_STRING=PJSIP)
exten => 2,n,Set(DIAL_TRUNK=2)
exten => 2,n,Goto(ext-trunk,tdial,1)
exten => tdial,1,Set(OUTBOUND_GROUP=OUT_${DIAL_TRUNK})
exten => tdial,n,GotoIf($["${OUTMAXCHANS_${DIAL_TRUNK}}" = ""]?nomax)
exten => tdial,n,GotoIf($[${GROUP_COUNT(OUT_${DIAL_TRUNK})} >= ${OUTMAXCHANS_${DIAL_TRUNK}}]?hangit)
exten => tdial,n(nomax),ExecIf($["${CALLINGPRES_SV}" != ""]?Set(CALLERPRES()=${CALLINGPRES_SV}))
exten => tdial,n,Set(DIAL_NUMBER=${FROM_DID})
exten => tdial,n,GosubIf($["${PREFIX_TRUNK_${DIAL_TRUNK}}" != ""]?sub-flp-${DIAL_TRUNK},s,1())
exten => tdial,n,Set(OUTNUM=${OUTPREFIX_${DIAL_TRUNK}}${DIAL_NUMBER})
exten => tdial,n,Set(DIAL_TRUNK_OPTIONS=${IF($["${DB_EXISTS(TRUNK/${DIAL_TRUNK}/dialopts)}" = "1"]?${DB_RESULT}:${TRUNK_OPTIONS})})
exten => tdial,n,Dial(${TDIAL_STRING}/${OUTNUM}${TDIAL_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})
exten => tdial,n,Set(CALLERID(number)=${CALLERID(number):0:40})
exten => tdial,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten => tdial,n(hangit),Hangup
;--== end of [ext-trunk] ==--;
[macro-prepend-cid]
exten => s,1,GotoIf($["${RGPREFIX}" = ""]?REPCID)
exten => s,n,GotoIf($["${RGPREFIX}" != "${CALLERID(name):0:${LEN(${RGPREFIX})}}"]?REPCID)
exten => s,n,Set(CALLERID(name)=${CALLERID(name):${LEN(${RGPREFIX})}})
exten => s,n,Set(_RGPREFIX=)
exten => s,n(REPCID),Set(_RGPREFIX=${ARG1})
exten => s,n,Set(CALLERID(name)=${RGPREFIX}${CALLERID(name)})
;--== end of [macro-prepend-cid] ==--;
[outbound-allroutes]
include => outrt-1 ; Normal
exten => foo,1,Noop(bar)
;--== end of [outbound-allroutes] ==--;
[outrt-1] ; Normal
exten => _+49X.,1,Macro(user-callerid,LIMIT,EXTERNAL,)
exten => _+49X.,n,Gosub(sub-record-check,s,1(out,${EXTEN},dontcare))
exten => _+49X.,n,ExecIf($[ "${CALLEE_ACCOUNCODE}" != "" ] ?Set(CDR(accountcode)=${CALLEE_ACCOUNCODE}))
exten => _+49X.,n,Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})})
exten => _+49X.,n,ExecIf($["${KEEPCID}"!="TRUE" & ${LEN(${DB(AMPUSER/${AMPUSER}/outboundcid)})}=0 & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE="Kay" <04074041501>))
exten => _+49X.,n,Set(_NODEST=)
exten => _+49X.,n,Macro(dialout-trunk,2,${EXTEN},,off)
exten => _+49X.,n,Macro(outisbusy,)
exten => _0Z.,1,Macro(user-callerid,LIMIT,EXTERNAL,)
exten => _0Z.,n,Gosub(sub-record-check,s,1(out,${EXTEN},dontcare))
exten => _0Z.,n,ExecIf($[ "${CALLEE_ACCOUNCODE}" != "" ] ?Set(CDR(accountcode)=${CALLEE_ACCOUNCODE}))
exten => _0Z.,n,Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})})
exten => _0Z.,n,ExecIf($["${KEEPCID}"!="TRUE" & ${LEN(${DB(AMPUSER/${AMPUSER}/outboundcid)})}=0 & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE="Kay" <04074041501>))
exten => _0Z.,n,Set(_NODEST=)
exten => _0Z.,n,Macro(dialout-trunk,2,${EXTEN},,off)
exten => _0Z.,n,Macro(outisbusy,)
exten => _Z.,1,Macro(user-callerid,LIMIT,EXTERNAL,)
exten => _Z.,n,Gosub(sub-record-check,s,1(out,${EXTEN},dontcare))
exten => _Z.,n,ExecIf($[ "${CALLEE_ACCOUNCODE}" != "" ] ?Set(CDR(accountcode)=${CALLEE_ACCOUNCODE}))
exten => _Z.,n,Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})})
exten => _Z.,n,ExecIf($["${KEEPCID}"!="TRUE" & ${LEN(${DB(AMPUSER/${AMPUSER}/outboundcid)})}=0 & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE="Kay" <04074041501>))
exten => _Z.,n,Set(_NODEST=)
exten => _Z.,n,Macro(dialout-trunk,2,${EXTEN},,off)
exten => _Z.,n,Macro(outisbusy,)
;--== end of [outrt-1] ==--;
[app-blackhole]
exten => hangup,1,Noop(Blackhole Dest: Hangup)
exten => hangup,n,Hangup
exten => zapateller,1,Noop(Blackhole Dest: Play SIT Tone)
exten => zapateller,n,Answer
exten => zapateller,n,Zapateller()
exten => musiconhold,1,Noop(Blackhole Dest: Put caller on hold forever)
exten => musiconhold,n,Answer
exten => musiconhold,n,MusicOnHold()
exten => congestion,1,Noop(Blackhole Dest: Congestion)
exten => congestion,n,Playtones(congestion)
exten => congestion,n,Congestion(20)
exten => congestion,n,Hangup
exten => busy,1,Noop(Blackhole Dest: Busy)
exten => busy,n,Busy(20)
exten => busy,n,Hangup
exten => ring,1,Noop(Blackhole Dest: Ring)
exten => ring,n,Answer
exten => ring,n,Playtones(ring)
exten => ring,n,Wait(300)
exten => ring,n,Hangup
;--== end of [app-blackhole] ==--;
[bad-number]
exten => _X.,1,ResetCDR()
exten => _X.,n,NoCDR()
exten => _X.,n,Progress
exten => _X.,n,Wait(1)
exten => _X.,n,Progress
exten => _X.,n,Playback(silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer)
exten => _X.,n,Wait(1)
exten => _X.,n,Congestion(20)
exten => _X.,n,Hangup
;--== end of [bad-number] ==--;
[sub-presencestate-display]
exten => s,1,Goto(state-${TOLOWER(${PRESENCE_STATE(CustomPresence:${ARG1},value)})},1)
exten => state-available,1,Set(PRESENCESTATE_DISPLAY=(Available))
exten => state-available,n,Return()
exten => state-chat,1,Set(PRESENCESTATE_DISPLAY=(Chatty))
exten => state-chat,n,Return()
exten => state-away,1,Set(PRESENCESTATE_DISPLAY=(Away))
exten => state-away,n,Return()
exten => state-dnd,1,Set(PRESENCESTATE_DISPLAY=(DND))
exten => state-dnd,n,Return()
exten => state-xa,1,Set(PRESENCESTATE_DISPLAY=(Extended Away))
exten => state-xa,n,Return()
exten => state-unavailable,1,Set(PRESENCESTATE_DISPLAY=(Unavailable))
exten => state-unavailable,n,Return()
exten => _state-.,1,Set(PRESENCESTATE_DISPLAY=)
exten => _state-.,n,Return()
exten => state-,1,Set(PRESENCESTATE_DISPLAY=)
exten => state-,n,Return()
;--== end of [sub-presencestate-display] ==--;
[macro-confirm]
exten => s,1,Set(LOOPCOUNT=0)
exten => s,n,Set(__MACRO_RESULT=ABORT)
exten => s,n,Set(MSG1=${IF($["${ARG1}${ALT_CONFIRM_MSG}"=""]?incoming-call-1-accept-2-decline:${IF($[${LEN(${ALT_CONFIRM_MSG})}>0]?${ALT_CONFIRM_MSG}:${ARG1})})})
exten => s,n(start),Background(${MSG1},m,${CHANNEL(language)},macro-confirm)
exten => s,n,Read(INPUT,,1,,,4)
exten => s,n,GotoIf($[${LEN(${INPUT})} > 0]?${INPUT},1:t,1)
exten => 1,1,GotoIf($["${DB_EXISTS(RG/${ARG3}/${UNIQCHAN})}"="0" & "${SHARED(ANSWER_STATUS,${FORCE_CONFIRM})}"=""]?toolate,1)
exten => 1,n,Noop(Deleting: RG/${ARG3}/${UNIQCHAN} ${DB_DELETE(RG/${ARG3}/${UNIQCHAN})})
exten => 1,n,Macro(blkvm-clr,)
exten => 1,n,Set(SHARED(ANSWER_STATUS,${FORCE_CONFIRM})=)
exten => 1,n(skip),Set(__MACRO_RESULT=)
exten => 1,n,ExecIf($[("${MOHCLASS}"!="default") & ("${MOHCLASS}"!="")]?Set(CHANNEL(musicclass)=${MOHCLASS}))
exten => 1,n(exitopt1),MacroExit()
exten => 2,1,Goto(noanswer,1)
exten => 3,1,SayDigits(${CALLCONFIRMCID})
exten => 3,n,GotoIf($["${DB_EXISTS(RG/${ARG3}/${UNIQCHAN})}"="0" & "${SHARED(ANSWER_STATUS,${FORCE_CONFIRM})}"=""]?toolate,1:s,start)
exten => t,1,GotoIf($["${DB_EXISTS(RG/${ARG3}/${UNIQCHAN})}"="0" & "${SHARED(ANSWER_STATUS,${FORCE_CONFIRM})}"=""]?toolate,1)
exten => t,n,Set(LOOPCOUNT=$[ ${LOOPCOUNT} + 1 ])
exten => t,n,GotoIf($[ ${LOOPCOUNT} < 5 ]?s,start:noanswer,1)
exten => _X,1,Background(invalid,m,${CHANNEL(language)},macro-confirm)
exten => _X,n,GotoIf($["${DB_EXISTS(RG/${ARG3}/${UNIQCHAN})}"="0" | "${SHARED(ANSWER_STATUS,${FORCE_CONFIRM})}"=""]?toolate,1)
exten => _X,n,Set(LOOPCOUNT=$[ ${LOOPCOUNT} + 1 ])
exten => _X,n,GotoIf($[ ${LOOPCOUNT} < 5 ]?s,start:noanswer,1)
exten => noanswer,1,Set(__MACRO_RESULT=ABORT)
exten => noanswer,n(exitnoanswer),MacroExit()
exten => toolate,1,Set(MSG2=${IF($["foo${ARG2}" != "foo"]?${ARG2}:"incoming-call-no-longer-avail")})
exten => toolate,n,Playback(${MSG2})
exten => toolate,n,Set(__MACRO_RESULT=ABORT)
exten => toolate,n(exittoolate),MacroExit()
exten => h,1,Macro(hangupcall,)
;--== end of [macro-confirm] ==--;
[macro-auto-confirm]
exten => s,1,Set(__MACRO_RESULT=)
exten => s,n,Set(CFIGNORE=)
exten => s,n,Set(MASTER_CHANNEL(CFIGNORE)=)
exten => s,n,Set(FORWARD_CONTEXT=from-internal)
exten => s,n,Set(MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal)
exten => s,n,Macro(blkvm-clr,)
exten => s,n,Noop(Deleting: RG/${ARG1}/${UNIQCHAN} ${DB_DELETE(RG/${ARG1}/${UNIQCHAN})})
exten => s,n,ExecIf($[!${REGEX("[^0-9]" ${DIALEDPEERNUMBER})} && "${DB(AMPUSER/${AMPUSER}/cidname)}" != ""]?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=${DIALEDPEERNUMBER}))
exten => s,n,ExecIf($[!${REGEX("[^0-9]" ${DIALEDPEERNUMBER})} && "${DB(AMPUSER/${AMPUSER}/cidname)}" != ""]?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=${DB(AMPUSER/${DIALEDPEERNUMBER}/cidname)}))
;--== end of [macro-auto-confirm] ==--;
[macro-auto-blkvm]
exten => s,1,Set(__MACRO_RESULT=)
exten => s,n,Set(CFIGNORE=)
exten => s,n,Set(MASTER_CHANNEL(CFIGNORE)=)
exten => s,n,Set(FORWARD_CONTEXT=from-internal)
exten => s,n,Set(MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal)
exten => s,n,Macro(blkvm-clr,)
exten => s,n,ExecIf($[!${REGEX("[^0-9]" ${DIALEDPEERNUMBER})} && "${DB(AMPUSER/${AMPUSER}/cidname)}" != ""]?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=${DIALEDPEERNUMBER}))
exten => s,n,ExecIf($[!${REGEX("[^0-9]" ${DIALEDPEERNUMBER})} && "${DB(AMPUSER/${AMPUSER}/cidname)}" != ""]?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=${DB(AMPUSER/${DIALEDPEERNUMBER}/cidname)}))
;--== end of [macro-auto-blkvm] ==--;
[sub-pincheck]
exten => s,1,Authenticate(${ARG3},)
exten => s,n,ResetCDR()
exten => s,n,Return()
;--== end of [sub-pincheck] ==--;
[macro-dialout-trunk]
exten => s,1,Set(DIAL_TRUNK=${ARG1})
exten => s,n,GosubIf($[$["${ARG3}" != ""] & $["${DB(AMPUSER/${AMPUSER}/pinless)}" != "NOPASSWD"]]?sub-pincheck,s,1())
exten => s,n,GotoIf($["x${OUTDISABLE_${DIAL_TRUNK}}" = "xon"]?disabletrunk,1)
exten => s,n,Set(DIAL_NUMBER=${ARG2})
exten => s,n,Set(DIAL_TRUNK_OPTIONS=${DIAL_OPTIONS})
exten => s,n,Set(OUTBOUND_GROUP=OUT_${DIAL_TRUNK})
exten => s,n,GotoIf($["${OUTMAXCHANS_${DIAL_TRUNK}}foo" = "foo"]?nomax)
exten => s,n,GotoIf($[ ${GROUP_COUNT(OUT_${DIAL_TRUNK})} >= ${OUTMAXCHANS_${DIAL_TRUNK}} ]?chanfull)
exten => s,n(nomax),GotoIf($["${INTRACOMPANYROUTE}" = "YES"]?skipoutcid)
exten => s,n,Set(DIAL_TRUNK_OPTIONS=${IF($["${DB_EXISTS(TRUNK/${DIAL_TRUNK}/dialopts)}" = "1"]?${DB_RESULT}:${TRUNK_OPTIONS})})
exten => s,n,Macro(outbound-callerid,${DIAL_TRUNK})
exten => s,n(skipoutcid),GosubIf($["${PREFIX_TRUNK_${DIAL_TRUNK}}" != ""]?sub-flp-${DIAL_TRUNK},s,1())
exten => s,n,Set(OUTNUM=${OUTPREFIX_${DIAL_TRUNK}}${DIAL_NUMBER})
exten => s,n,Set(custom=${CUT(OUT_${DIAL_TRUNK},:,1)})
exten => s,n,ExecIf($["${MOHCLASS}"!="default" & "${MOHCLASS}"!="" & "${FORCE_CONFIRM}"="" ]?Set(DIAL_TRUNK_OPTIONS=M(setmusic^${MOHCLASS})${DIAL_TRUNK_OPTIONS}))
exten => s,n,ExecIf($["${FORCE_CONFIRM}"!="" ]?Set(DIAL_TRUNK_OPTIONS=${DIAL_TRUNK_OPTIONS}M(confirm)))
exten => s,n(gocall),Macro(dialout-trunk-predial-hook,)
exten => s,n,GotoIf($["${PREDIAL_HOOK_RET}" = "BYPASS"]?bypass,1)
exten => s,n,ExecIf($["${DB(AMPUSER/${AMPUSER}/cidname)}" != ""]?Set(CONNECTEDLINE(num,i)=${DIAL_NUMBER}))
exten => s,n,ExecIf($["${DB(AMPUSER/${AMPUSER}/cidname)}" != ""]?Set(CONNECTEDLINE(name,i)=CID:${CALLERID(number)}))
exten => s,n,GotoIf($["${custom}" = "AMP"]?customtrunk)
exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})
exten => s,n,Noop(Dial failed for some reason with DIALSTATUS = ${DIALSTATUS} and HANGUPCAUSE = ${HANGUPCAUSE})
exten => s,n,GotoIf($["${ARG4}" = "on"]?continue,1:s-${DIALSTATUS},1)
exten => s,n(customtrunk),Set(pre_num=${CUT(OUT_${DIAL_TRUNK},$,1)})
exten => s,n,Set(the_num=${CUT(OUT_${DIAL_TRUNK},$,2)})
exten => s,n,Set(post_num=${CUT(OUT_${DIAL_TRUNK},$,3)})
exten => s,n,GotoIf($["${the_num}" = "OUTNUM"]?outnum:skipoutnum)
exten => s,n(outnum),Set(the_num=${OUTNUM})
exten => s,n(skipoutnum),Dial(${pre_num:4}${the_num}${post_num},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})
exten => s,n,Noop(Dial failed for some reason with DIALSTATUS = ${DIALSTATUS} and HANGUPCAUSE = ${HANGUPCAUSE})
exten => s,n,GotoIf($["${ARG4}" = "on"]?continue,1:s-${DIALSTATUS},1)
exten => s,n(chanfull),Noop(max channels used up)
exten => s-BUSY,1,Noop(Dial failed due to trunk reporting BUSY - giving up)
exten => s-BUSY,n,Playtones(busy)
exten => s-BUSY,n,Busy(20)
exten => s-ANSWER,1,Noop(Call successfully answered - Hanging up now)
exten => s-ANSWER,n,Macro(hangupcall,)
exten => s-NOANSWER,1,Noop(Dial failed due to trunk reporting NOANSWER - giving up)
exten => s-NOANSWER,n,Progress
exten => s-NOANSWER,n,Playback(number-not-answering,noanswer)
exten => s-NOANSWER,n,Congestion(20)
exten => s-INVALIDNMBR,1,Noop(Dial failed due to trunk reporting Address Incomplete - giving up)
exten => s-INVALIDNMBR,n,Progress
exten => s-INVALIDNMBR,n,Playback(ss-noservice,noanswer)
exten => s-INVALIDNMBR,n,Busy(20)
exten => s-CHANGED,1,Noop(Dial failed due to trunk reporting Number Changed - giving up)
exten => s-CHANGED,n,Playtones(busy)
exten => s-CHANGED,n,Busy(20)
exten => _s-.,1,Set(RC=${IF($[${ISNULL(${HANGUPCAUSE})}]?0:${HANGUPCAUSE})})
exten => _s-.,n,Goto(${RC},1)
exten => 17,1,Goto(s-BUSY,1)
exten => 18,1,Goto(s-NOANSWER,1)
exten => 22,1,Goto(s-CHANGED,1)
exten => 23,1,Goto(s-CHANGED,1)
exten => 1,1,Goto(s-INVALIDNMBR,1)
exten => 28,1,Goto(s-INVALIDNMBR,1)
exten => _X,1,Goto(continue,1)
exten => _X.,1,Goto(continue,1)
exten => continue,1(noreport),Noop(TRUNK Dial failed due to ${DIALSTATUS} HANGUPCAUSE: ${HANGUPCAUSE} - failing through to other trunks)
exten => continue,n,Set(CALLERID(number)=${AMPUSER})
exten => disabletrunk,1,Noop(TRUNK: ${OUT_${DIAL_TRUNK}} DISABLED - falling through to next trunk)
exten => bypass,1,Noop(TRUNK: ${OUT_${DIAL_TRUNK}} BYPASSING because dialout-trunk-predial-hook)
exten => h,1,Macro(hangupcall,)
;--== end of [macro-dialout-trunk] ==--;
[macro-privacy-mgr]
exten => s,1,Set(KEEPCID=${CALLERID(num)})
exten => s,n,Set(TESTCID=${IF($["${CALLERID(num):0:1}"="+"]?${MATH(1+${CALLERID(num):1})}:${MATH(1+${CALLERID(num)})})})
exten => s,n,ExecIf($[${LEN(${TESTCID})}=0]?Set(CALLERID(num)=))
exten => s,n,PrivacyManager(${ARG1},${ARG2})
exten => s,n,GotoIf($["${PRIVACYMGRSTATUS}"="FAILED"]?fail)
exten => s,n,GosubIf($["${CALLED_BLACKLIST}"="1"]?app-blacklist-check,s,1())
exten => s,n,Set(CALLERID(num-pres)=allowed_passed_screen)
exten => s,n,MacroExit()
exten => s,n(fail),Noop(STATUS: ${PRIVACYMGRSTATUS} CID: ${CALLERID(num)} ${CALLERID(name)} CALLPRES: ${CALLLINGPRES})
exten => s,n,Playback(sorry-youre-having-problems&goodbye)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(20)
exten => h,1,Hangup
;--== end of [macro-privacy-mgr] ==--;
[macro-user-callerid]
exten => s,1,Set(TOUCH_MONITOR=${UNIQUEID})
exten => s,n,Set(AMPUSER=${IF($["${AMPUSER}" = ""]?${CALLERID(number)}:${AMPUSER})})
exten => s,n,GotoIf($["${CUT(CHANNEL,@,2):5:5}"="queue" | ${LEN(${AMPUSERCIDNAME})}]?report)
exten => s,n,ExecIf($["${REALCALLERIDNUM:1:2}" = ""]?Set(REALCALLERIDNUM=${CALLERID(number)}))
exten => s,n,Set(AMPUSER=${DB(DEVICE/${REALCALLERIDNUM}/user)})
exten => s,n,GotoIf($["${AMPUSER}" = "none"]?limit)
exten => s,n,Set(AMPUSERCIDNAME=${DB(AMPUSER/${AMPUSER}/cidname)})
exten => s,n,GotoIf($["${AMPUSERCIDNAME:1:2}" = ""]?report)
exten => s,n,Set(AMPUSERCID=${IF($["${ARG2}" != "EXTERNAL" & "${DB_EXISTS(AMPUSER/${AMPUSER}/cidnum)}" = "1"]?${DB_RESULT}:${AMPUSER})})
exten => s,n,Set(__DIAL_OPTIONS=${IF($["${DB_EXISTS(AMPUSER/${AMPUSER}/dialopts)}" = "1"]?${DB_RESULT}:${DIAL_OPTIONS})})
exten => s,n,Set(CALLERID(all)="${AMPUSERCIDNAME}" <${AMPUSERCID}>)
exten => s,n,GotoIf($["${ARG1}"="LIMIT" & ${LEN(${AMPUSER})} & ${DB_EXISTS(AMPUSER/${AMPUSER}/concurrency_limit)} & ${DB(AMPUSER/${AMPUSER}/concurrency_limit)}>0 & ${GROUP_COUNT(${AMPUSER}@concurrency_limit)}>=${DB(AMPUSER/${AMPUSER}/concurrency_limit)}]?limit)
exten => s,n,ExecIf($["${ARG1}"="LIMIT" & ${LEN(${AMPUSER})}]?Set(GROUP(concurrency_limit)=${AMPUSER}))
exten => s,n,GosubIf($[${LEN(${DB(AMPUSER/${AMPUSER}/ccss/cc_agent_policy)})} & "${DB(AMPUSER/${AMPUSER}/ccss/cc_agent_policy)}" != "never"]?sub-ccss,s,1(${MACRO_CONTEXT},${CALLERID(dnid)}))
exten => s,n,ExecIf($["${DB(AMPUSER/${AMPUSER}/language)}" != ""]?Set(CHANNEL(language)=${DB(AMPUSER/${AMPUSER}/language)}))
exten => s,n(report),GotoIf($[ "${ARG1}" = "SKIPTTL" | "${ARG1}" = "LIMIT" ]?continue)
exten => s,n,ExecIf($["${CALLEE_ACCOUNCODE}" = ""]?Set(__CALLEE_ACCOUNCODE=${DB(AMPUSER/${IF($["${MACRO_CONTEXT}"="macro-exten-vm"]?${ARG2}:${MACRO_EXTEN})}/accountcode)}))
exten => s,n(report2),Set(__TTL=${IF($["foo${TTL}" = "foo"]?64:$[ ${TTL} - 1 ])})
exten => s,n,GotoIf($[ ${TTL} > 0 ]?continue)
exten => s,n,Wait(${RINGTIMER})
exten => s,n,Answer
exten => s,n,Wait(1)
exten => s,n,Playback(im-sorry&an-error-has-occurred&with&call-forwarding)
exten => s,n,Macro(hangupcall,)
exten => s,n(limit),Answer
exten => s,n,Wait(1)
exten => s,n,Playback(beep&im-sorry&your&simul-call-limit-reached&goodbye)
exten => s,n,Macro(hangupcall,)
exten => s,n,Congestion(20)
exten => s,n(continue),Set(CALLERID(number)=${CALLERID(number):0:40})
exten => s,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten => s,n,Set(CDR(cnum)=${CALLERID(num)})
exten => s,n,Set(CDR(cnam)=${CALLERID(name)})
exten => s,n,Set(CHANNEL(language)=${MASTER_CHANNEL(CHANNEL(language))})
exten => h,1,Macro(hangupcall,)
;--== end of [macro-user-callerid] ==--;
[macro-outbound-callerid]
exten => s,1,ExecIf($["${CALLINGPRES_SV}" != ""]?Set(CALLERPRES()=${CALLINGPRES_SV}))
exten => s,n,ExecIf($["${REALCALLERIDNUM:1:2}" = ""]?Set(REALCALLERIDNUM=${CALLERID(number)}))
exten => s,n(start),GotoIf($[ $["${REALCALLERIDNUM}" = ""] | $["${KEEPCID}" != "TRUE"] | $["${OUTKEEPCID_${ARG1}}" = "on"] ]?normcid)
exten => s,n,Set(USEROUTCID=${REALCALLERIDNUM})
exten => s,n,GotoIf($["foo${DB(AMPUSER/${REALCALLERIDNUM}/device)}" = "foo"]?bypass)
exten => s,n(normcid),Set(USEROUTCID=${DB(AMPUSER/${AMPUSER}/outboundcid)})
exten => s,n(bypass),Set(EMERGENCYCID=${DB(DEVICE/${REALCALLERIDNUM}/emergency_cid)})
exten => s,n,Set(TRUNKOUTCID=${OUTCID_${ARG1}})
exten => s,n,GotoIf($["${EMERGENCYROUTE:1:2}" = "" | "${EMERGENCYCID:1:2}" = ""]?trunkcid)
exten => s,n,Set(CALLERID(all)=${EMERGENCYCID})
exten => s,n,Set(CDR(outbound_cnum)=${CALLERID(num)})
exten => s,n,Set(CDR(outbound_cnam)=${CALLERID(name)})
exten => s,n(exit),MacroExit()
exten => s,n(trunkcid),ExecIf($[${LEN(${TRUNKOUTCID})} != 0]?Set(CALLERID(all)=${TRUNKOUTCID}))
exten => s,n(usercid),ExecIf($[${LEN(${USEROUTCID})} != 0]?Set(CALLERID(all)=${USEROUTCID}))
exten => s,n,ExecIf($[${LEN(${TRUNKCIDOVERRIDE})} != 0 | ${LEN(${FORCEDOUTCID_${ARG1}})} != 0]?Set(CALLERID(all)=${IF($[${LEN(${FORCEDOUTCID_${ARG1}})}=0]?${TRUNKCIDOVERRIDE}:${FORCEDOUTCID_${ARG1}})}))
exten => s,n(hidecid),ExecIf($["${CALLERID(name)}"="hidden"]?Set(CALLERPRES()=prohib_passed_screen))
exten => s,n,Set(CDR(outbound_cnum)=${CALLERID(num)})
exten => s,n,Set(CDR(outbound_cnam)=${CALLERID(name)})
;--== end of [macro-outbound-callerid] ==--;
[from-zaptel]
include => from-dahdi
exten => foo,1,Noop(bar)
;--== end of [from-zaptel] ==--;
[from-dahdi]
exten => _X.,1,Set(DID=${EXTEN})
exten => _X.,n,Goto(s,1)
exten => s,1,Noop(Entering from-dahdi with DID == ${DID})
exten => s,n,Ringing()
exten => s,n,Set(DID=${IF($["${DID}"= ""]?s:${DID})})
exten => s,n,Noop(DID is now ${DID})
exten => s,n,GotoIf($["${CHANNEL:0:5}"="DAHDI"]?dahdiok:checkzap)
exten => s,n(checkzap),GotoIf($["${CHANNEL:0:3}"="Zap"]?zapok:neither)
exten => s,n(neither),Goto(from-pstn,${DID},1)
exten => s,n,Macro(Hangupcall,dummy)
exten => s,n(dahdiok),Noop(Is a DAHDi Channel)
exten => s,n,Set(CHAN=${CHANNEL:6})
exten => s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten => s,n,Macro(from-dahdi-${CHAN},${DID},1)
exten => s,n,Noop(Returned from Macro from-dahdi-${CHAN})
exten => s,n,Goto(from-pstn,${DID},1)
exten => s,n(zapok),Noop(Is a Zaptel Channel)
exten => s,n,Set(CHAN=${CHANNEL:4})
exten => s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten => s,n,Macro(from-dahdi-${CHAN},${DID},1)
exten => s,n,Noop(Returned from Macro from-dahdi-${CHAN})
exten => s,n,Goto(from-pstn,${DID},1)
;--== end of [from-dahdi] ==--;
[macro-dial-confirm]
exten => s,1,Set(DB(RG/${ARG4}/${CHANNEL})=RINGING)
exten => s,n,Set(__UNIQCHAN=${CHANNEL})
exten => s,n,Set(USE_CONFIRMATION=TRUE)
exten => s,n,Set(RINGGROUP_INDEX=${ARG4})
exten => s,n,Set(FORCE_CONFIRM=)
exten => s,n,Set(ARG4=)
exten => s,n,Macro(dial,${ARG1},${ARG2},${ARG3})
exten => s,n,Noop(Deleting: RG/${RINGGROUP_INDEX}/${CHANNEL} ${DB_DELETE(RG/${RINGGROUP_INDEX}/${CHANNEL})})
exten => s,n,Set(USE_CONFIRMATION=)
exten => s,n,Set(RINGGROUP_INDEX=)
;--== end of [macro-dial-confirm] ==--;
[macro-setmusic]
exten => s,1,Set(CHANNEL(musicclass)=${ARG1})
;--== end of [macro-setmusic] ==--;
[macro-block-cf]
exten => _X.,1,Hangup
exten => s-ANSWER,1,Noop(Call successfully answered - Hanging up now)
exten => s-ANSWER,n,Macro(hangupcall,)
;--== end of [macro-block-cf] ==--;
[macro-vm]
exten => s,1,Macro(user-callerid,SKIPTTL)
exten => s,n,Set(VMGAIN=${IF($["foo${VM_GAIN}"!="foo"]?"g(${VM_GAIN})": )})
exten => s,n,Macro(blkvm-check,)
exten => s,n,GotoIf($["${GOSUB_RETVAL}" != "TRUE"]?vmx,1)
exten => s,n,Hangup
exten => vmx,1,Set(MEXTEN=${ARG1})
exten => vmx,n,Set(MMODE=${ARG2})
exten => vmx,n,Set(RETVM=${ARG3})
exten => vmx,n,Set(MODE=${IF($["${MMODE}"="BUSY"]?busy:unavail)})
exten => vmx,n,Macro(get-vmcontext,${MEXTEN})
exten => vmx,n,Set(MODE=${IF($[(${STAT(f,${ASTSPOOLDIR}/voicemail/${VMCONTEXT}/${MEXTEN}/temp.wav)} = 1) || (${STAT(f,${ASTSPOOLDIR}/voicemail/${VMCONTEXT}/${MEXTEN}/temp.WAV)} = 1)]?temp:${MODE})})
exten => vmx,n,Noop(MODE IS: ${MODE})
exten => vmx,n,GotoIf($["${DB(AMPUSER/${MEXTEN}/vmx/${MODE}/state)}" != "enabled"]?chknomsg)
exten => vmx,n,Set(VM_OPTS=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/vmxopts/timeout)}" = "1"]?${DB_RESULT}:${VM_OPTS})})
exten => vmx,n(chknomsg),GotoIf($["${MMODE}"="NOMESSAGE"]?s-${MMODE},1)
exten => vmx,n,GotoIf($["${MMODE}" != "DIRECTDIAL"]?notdirect)
exten => vmx,n,Set(MODE=${IF($["${REGEX("[b]" ${VM_DDTYPE})}" = "1"]?busy:${MODE})})
exten => vmx,n(notdirect),Noop(Checking if ext ${MEXTEN} is enabled: ${DB(AMPUSER/${MEXTEN}/vmx/${MODE}/state)})
exten => vmx,n,GotoIf($["${DB(AMPUSER/${MEXTEN}/vmx/${MODE}/state)}" != "enabled"]?s-${MMODE},1)
exten => vmx,n,GotoIf($[(${STAT(f,${ASTSPOOLDIR}/voicemail/${VMCONTEXT}/${MEXTEN}/${MODE}.wav)} = 0) && (${STAT(f,${ASTSPOOLDIR}/voicemail/${VMCONTEXT}/${MEXTEN}/${MODE}.WAV)} = 0)]?nofile)
exten => vmx,n,Set(VMX_TIMEOUT=2)
exten => vmx,n,Set(VMX_REPEAT=1)
exten => vmx,n,Set(VMX_LOOPS=1)
exten => vmx,n,Set(LOOPCOUNT=0)
exten => vmx,n,Set(VMX_REPEAT=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/repeat)}" = "1"]?${DB_RESULT}:${VMX_REPEAT})})
exten => vmx,n(vmxtime),Set(VMX_TIMEOUT=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/timeout)}" = "1"]?${DB_RESULT}:${VMX_TIMEOUT})})
exten => vmx,n(vmxloops),Set(VMX_LOOPS=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/loops)}" = "1"]?${DB_RESULT}:${VMX_LOOPS})})
exten => vmx,n(vmxanswer),Answer
exten => vmx,n(loopstart),Read(ACTION,${ASTSPOOLDIR}/voicemail/${VMCONTEXT}/${MEXTEN}/${MODE},1,skip,${VMX_REPEAT},${VMX_TIMEOUT})
exten => vmx,n,GotoIf($["${EXISTS(${ACTION})}" = "1"]?checkopt)
exten => vmx,n(noopt),Noop(Timeout: going to timeout dest)
exten => vmx,n,Set(VMX_OPTS=${DB(AMPUSER/${MEXTEN}/vmx/${MODE}/vmxopts/timeout)})
exten => vmx,n(chktime),GotoIf($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/timedest/ext)}" = "0"]?dotime)
exten => vmx,n,Set(VMX_TIMEDEST_EXT=${DB_RESULT})
exten => vmx,n,Set(VMX_TIMEDEST_CONTEXT=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/timedest/context)}" = "1"]?${DB_RESULT}:${VMX_TIMEDEST_CONTEXT})})
exten => vmx,n(timepri),Set(VMX_TIMEDEST_PRI=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/timedest/pri)}" = "1"]?${DB_RESULT}:${VMX_TIMEDEST_PRI})})
exten => vmx,n(dotime),Goto(${VMX_TIMEDEST_CONTEXT},${VMX_TIMEDEST_EXT},${VMX_TIMEDEST_PRI})
exten => vmx,n(checkopt),GotoIf($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/${ACTION}/ext)}" = "1"]?doopt)
exten => vmx,n,GotoIf($["${ACTION}" = "0"]?o,1)
exten => vmx,n,GotoIf($["${ACTION}" = "*"]?adef,1)
exten => vmx,n,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
exten => vmx,n,GotoIf($["${LOOPCOUNT}" > "${VMX_LOOPS}"]?toomany)
exten => vmx,n,Playback(pm-invalid-option&please-try-again)
exten => vmx,n,Goto(loopstart)
exten => vmx,n(toomany),Noop(Too Many invalid entries, got to invalid dest)
exten => vmx,n,Set(VMX_OPTS=${VMX_OPTS_LOOP})
exten => vmx,n,Set(VMX_OPTS=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/vmxopts/loops)}" = "1"]?${DB_RESULT}:${VMX_OPTS})})
exten => vmx,n(chkloop),GotoIf($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/loopdest/ext)}" = "0"]?doloop)
exten => vmx,n,Set(VMX_LOOPDEST_EXT=${DB_RESULT})
exten => vmx,n,Set(VMX_LOOPDEST_CONTEXT=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/loopdest/context)}" = "1"]?${DB_RESULT}:${VMX_LOOPDEST_CONTEXT})})
exten => vmx,n(looppri),Set(VMX_LOOPDEST_PRI=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/loopdest/pri)}" = "1"]?${DB_RESULT}:${VMX_LOOPDEST_PRI})})
exten => vmx,n(doloop),Goto(${VMX_LOOPDEST_CONTEXT},${VMX_LOOPDEST_EXT},${VMX_LOOPDEST_PRI})
exten => vmx,n(doopt),Noop(Got a valid option: ${DB_RESULT})
exten => vmx,n,Set(VMX_EXT=${DB_RESULT})
exten => vmx,n,GotoIf($["${VMX_EXT}" != "dovm"]?getdest)
exten => vmx,n(vmxopts),Set(VMX_OPTS=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/vmxopts/dovm)}" = "1"]?${DB_RESULT}:${VMX_OPTS_DOVM})})
exten => vmx,n(vmxdovm),Goto(dovm,1)
exten => vmx,n(getdest),Set(VMX_CONTEXT=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/${ACTION}/context)}" = "1"]?${DB_RESULT}:${VMX_CONTEXT})})
exten => vmx,n(vmxpri),Set(VMX_PRI=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/${ACTION}/pri)}" = "1"]?${DB_RESULT}:${VMX_PRI})})
exten => vmx,n(vmxgoto),Goto(${VMX_CONTEXT},${VMX_EXT},${VMX_PRI})
exten => vmx,n(nofile),Noop(File for mode: ${MODE} does not exist, SYSTEMSTATUS: ${SYSTEMSTATUS}, going to normal voicemail)
exten => vmx,n,Goto(s-${MMODE},1)
exten => vmx,n(tmpgreet),Noop(Temporary Greeting Detected, going to normal voicemail)
exten => vmx,n,Goto(s-${MMODE},1)
exten => dovm,1,Noop(VMX Timeout - go to voicemail)
exten => dovm,n,VoiceMail(${MEXTEN}@${VMCONTEXT},${VMX_OPTS}${VMGAIN})
exten => dovm,n,Goto(exit-${VMSTATUS},1)
exten => s-BUSY,1,Noop(BUSY voicemail)
exten => s-BUSY,n,Macro(get-vmcontext,${MEXTEN})
exten => s-BUSY,n,VoiceMail(${MEXTEN}@${VMCONTEXT},${VM_OPTS}b${VMGAIN})
exten => s-BUSY,n,Goto(exit-${VMSTATUS},1)
exten => s-NOMESSAGE,1,Noop(NOMESSAGE (beep only) voicemail)
exten => s-NOMESSAGE,n,Macro(get-vmcontext,${MEXTEN})
exten => s-NOMESSAGE,n,VoiceMail(${MEXTEN}@${VMCONTEXT},s${VM_OPTS}${VMGAIN})
exten => s-NOMESSAGE,n,Goto(exit-${VMSTATUS},1)
exten => s-INSTRUCT,1,Noop(NOMESSAGE (beeb only) voicemail)
exten => s-INSTRUCT,n,Macro(get-vmcontext,${MEXTEN})
exten => s-INSTRUCT,n,VoiceMail(${MEXTEN}@${VMCONTEXT},${VM_OPTS}${VMGAIN})
exten => s-INSTRUCT,n,Goto(exit-${VMSTATUS},1)
exten => s-DIRECTDIAL,1,Noop(DIRECTDIAL voicemail)
exten => s-DIRECTDIAL,n,Macro(get-vmcontext,${MEXTEN})
exten => s-DIRECTDIAL,n,VoiceMail(${MEXTEN}@${VMCONTEXT},${VM_OPTS}${VM_DDTYPE}${VMGAIN})
exten => s-DIRECTDIAL,n,Goto(exit-${VMSTATUS},1)
exten => _s-.,1,Macro(get-vmcontext,${MEXTEN})
exten => _s-.,n,VoiceMail(${MEXTEN}@${VMCONTEXT},${VM_OPTS}u${VMGAIN})
exten => _s-.,n,Goto(exit-${VMSTATUS},1)
exten => o,1,Playback(one-moment-please)
exten => o,n,GotoIf($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/0/ext)}" = "0"]?doopdef)
exten => o,n,Set(VMX_OPDEST_EXT=${DB_RESULT})
exten => o,n(opcontext),Set(VMX_OPDEST_CONTEXT=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/0/context)}" = "1"]?${DB_RESULT}:${VMX_CONTEXT})})
exten => o,n(oppri),Set(VMX_OPDEST_PRI=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/0/pri)}" = "1"]?${DB_RESULT}:${VMX_PRI})})
exten => o,n,Goto(${VMX_OPDEST_CONTEXT},${VMX_OPDEST_EXT},${VMX_OPDEST_PRI})
exten => o,n(doopdef),GotoIf($["x${OPERATOR_XTN}"="x"]?nooper:from-internal,${OPERATOR_XTN},1)
exten => o,n(nooper),GotoIf($["x${FROM_DID}"="x"]?nodid)
exten => o,n,Dial(Local/${FROM_DID}@from-pstn,)
exten => o,n,Macro(hangup,)
exten => o,n(nodid),Dial(Local/s@from-pstn,)
exten => o,n,Macro(hangup,)
exten => a,1,Macro(get-vmcontext,${MEXTEN})
exten => a,n,GotoIf($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/*/ext)}" = "0"]?adef,1)
exten => a,n,Set(VMX_ADEST_EXT=${DB_RESULT})
exten => a,n(acontext),Set(VMX_ADEST_CONTEXT=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/*/context)}" = "1"]?${DB_RESULT}:${VMX_CONTEXT})})
exten => a,n(apri),Set(VMX_ADEST_PRI=${IF($["${DB_EXISTS(AMPUSER/${MEXTEN}/vmx/${MODE}/*/pri)}" = "1"]?${DB_RESULT}:${VMX_PRI})})
exten => a,n,Goto(${VMX_ADEST_CONTEXT},${VMX_ADEST_EXT},${VMX_ADEST_PRI})
exten => adef,1,VoiceMailMain(${MEXTEN}@${VMCONTEXT})
exten => adef,n,GotoIf($["${RETVM}" = "RETURN"]?exit-RETURN,1)
exten => adef,n,Hangup
exten => exit-FAILED,1,Playback(im-sorry&an-error-has-occurred)
exten => exit-FAILED,n,GotoIf($["${RETVM}" = "RETURN"]?exit-RETURN,1)
exten => exit-FAILED,n,Hangup
exten => exit-SUCCESS,1,GotoIf($["${RETVM}" = "RETURN"]?exit-RETURN,1)
exten => exit-SUCCESS,n,Playback(goodbye)
exten => exit-SUCCESS,n,Hangup
exten => exit-USEREXIT,1,GotoIf($["${RETVM}" = "RETURN"]?exit-RETURN,1)
exten => exit-USEREXIT,n,Playback(goodbye)
exten => exit-USEREXIT,n,Hangup
exten => exit-RETURN,1,Noop(Returning From Voicemail because macro)
exten => t,1,Hangup
;--== end of [macro-vm] ==--;
[macro-exten-vm]
exten => s,1,Macro(user-callerid,)
exten => s,n,Set(RingGroupMethod=none)
exten => s,n,Set(__EXTTOCALL=${ARG2})
exten => s,n,Set(__PICKUPMARK=${ARG2})
exten => s,n,Set(RT=${IF($["${ARG1}"!="novm" | "${DB(CFU/${EXTTOCALL})}"!="" | "${DB(CFB/${EXTTOCALL})}"!="" | "${ARG3}"="1" | "${ARG4}"="1" | "${ARG5}"="1"]?${RINGTIMER}:)})
exten => s,n,ExecIf($[$["${REDIRECTING(reason)}" = "send_to_vm" | "${SIP_HEADER(X-Digium-Call-Feature)}" = "feature_send_to_vm"] & "${ARG1}" != "novm"]?Macro(vm,${ARG1},DIRECTDIAL,${IVR_RETVM}))
exten => s,n,ExecIf($[$["${REDIRECTING(reason)}" = "send_to_vm" | "${SIP_HEADER(X-Digium-Call-Feature)}" = "feature_send_to_vm"] & "${ARG1}" != "novm"]?MacroExit())
exten => s,n(checkrecord),Gosub(sub-record-check,s,1(exten,${EXTTOCALL},dontcare))
exten => s,n,GotoIf($["${AMPUSER}"=""|${LEN(${FROM_DID})}|"${DB(AMPUSER/${EXTTOCALL}/answermode)}"!="intercom"|${LEN(${BLINDTRANSFER})}|"${EXTENSION_STATE(${EXTTOCALL})}"!="NOT_INUSE"|"${CC_RECALL}"!=""]?macrodial)
exten => s,n,Set(INTERCOM_EXT_DOPTIONS=${DIAL_OPTIONS})
exten => s,n,Set(INTERCOM_RETURN=TRUE)
exten => s,n,Gosub(ext-intercom,*80${EXTTOCALL},1())
exten => s,n,Set(INTERCOM_RETURN=)
exten => s,n,Set(INTERCOM_EXT_DOPTIONS=)
exten => s,n(macrodial),GosubIf($["${INTERCOM_CALL}"="TRUE" & ${LEN(${BLINDTRANSFER})}]?clrheader,1())
exten => s,n,Macro(dial-one,${RT},${DIAL_OPTIONS},${EXTTOCALL})
exten => s,n,Set(SV_DIALSTATUS=${DIALSTATUS})
exten => s,n(calldocfu),GosubIf($[("${SV_DIALSTATUS}"="NOANSWER"|"${SV_DIALSTATUS}"="CHANUNAVAIL") & "${DB(CFU/${EXTTOCALL})}"!="" & "${SCREEN}"=""]?docfu,1())
exten => s,n(calldocfb),GosubIf($["${SV_DIALSTATUS}"="BUSY" & "${DB(CFB/${EXTTOCALL})}"!="" & "${SCREEN}"=""]?docfb,1())
exten => s,n,Set(DIALSTATUS=${SV_DIALSTATUS})
exten => s,n,ExecIf($[("${DIALSTATUS}"="NOANSWER"&"${ARG3}"="1")|("${DIALSTATUS}"="BUSY"&"${ARG4}"="1")|("${DIALSTATUS}"="CHANUNAVAIL"&"${ARG5}"="1")]?MacroExit())
exten => s,n,GotoIf($["${ARG1}"="novm"]?s-${DIALSTATUS},1)
exten => s,n,Macro(vm,${ARG1},${DIALSTATUS},${IVR_RETVM})
exten => docfu,1(docfu),ExecIf($["${DB(AMPUSER/${EXTTOCALL}/cfringtimer)}"="-1"|("${ARG1}"="novm"&"${ARG3}"="1")]?StackPop())
exten => docfu,n,GotoIf($["${DB(AMPUSER/${EXTTOCALL}/cfringtimer)}"="-1"|("${ARG1}"="novm"&"${ARG3}"="1")]?from-internal,${DB(CFU/${EXTTOCALL})},1)
exten => docfu,n,Set(RTCF=${IF($["${DB(AMPUSER/${EXTTOCALL}/cfringtimer)}"="0"]?${RT}:${DB(AMPUSER/${EXTTOCALL}/cfringtimer)})})
exten => docfu,n,ExecIf($["${DIRECTION}" = "INBOUND"]?Set(DIAL_OPTIONS=${REPLACE(DIAL_OPTIONS,T)}I))
exten => docfu,n,Dial(Local/${DB(CFU/${EXTTOCALL})}@from-internal/n,${RTCF},${DIAL_OPTIONS})
exten => docfu,n,Return()
exten => docfb,1(docfu),ExecIf($["${DB(AMPUSER/${EXTTOCALL}/cfringtimer)}"="-1"|("${ARG1}"="novm"&"${ARG4}"="1")]?StackPop())
exten => docfb,n,GotoIf($["${DB(AMPUSER/${EXTTOCALL}/cfringtimer)}"="-1"|("${ARG1}"="novm"&"${ARG4}"="1")]?from-internal,${DB(CFB/${EXTTOCALL})},1)
exten => docfb,n,Set(RTCF=${IF($["${DB(AMPUSER/${EXTTOCALL}/cfringtimer)}"="0"]?${RT}:${DB(AMPUSER/${EXTTOCALL}/cfringtimer)})})
exten => docfb,n,ExecIf($["${DIRECTION}" = "INBOUND"]?Set(DIAL_OPTIONS=${REPLACE(DIAL_OPTIONS,T)}I))
exten => docfb,n,Dial(Local/${DB(CFB/${EXTTOCALL})}@from-internal/n,${RTCF},${DIAL_OPTIONS})
exten => docfb,n,Return()
exten => clrheader,1,ExecIf($[${LEN(${SIPURI})}&"${SIPURI}"="${SIP_URI_OPTIONS}"]?Set(SIP_URI_OPTIONS=))
exten => clrheader,n,ExecIf($[${LEN(${ALERTINFO})}]?SIPRemoveHeader(${ALERTINFO}))
exten => clrheader,n,ExecIf($[${LEN(${CALLINFO})}]?SIPRemoveHeader(${CALLINFO}))
exten => clrheader,n,Return()
exten => s-BUSY,1,GotoIf($["${IVR_RETVM}"="RETURN" & "${IVR_CONTEXT}"!=""]?exit,1)
exten => s-BUSY,n,Playtones(busy)
exten => s-BUSY,n,Busy(20)
exten => _s-!,1,GotoIf($["${IVR_RETVM}"="RETURN" & "${IVR_CONTEXT}"!=""]?exit,1)
exten => _s-!,n,Playtones(congestion)
exten => _s-!,n,Congestion(10)
exten => exit,1,Playback(beep&line-busy-transfer-menu&silence/1)
exten => exit,n,MacroExit()
;--== end of [macro-exten-vm] ==--;
[macro-simple-dial]
exten => s,1,Set(__EXTTOCALL=${ARG1})
exten => s,n,Set(RT=${ARG2})
exten => s,n,Set(CFUEXT=${DB(CFU/${EXTTOCALL})})
exten => s,n,Set(CFBEXT=${DB(CFB/${EXTTOCALL})})
exten => s,n,Set(CWI_TMP=${CWIGNORE})
exten => s,n(macrodial),Macro(dial-one,${RT},${DIAL_OPTIONS},${EXTTOCALL})
exten => s,n,Set(__CWIGNORE=${CWI_TMP})
exten => s,n,Set(PR_DIALSTATUS=${DIALSTATUS})
exten => s,n(calldocfu),GosubIf($["${PR_DIALSTATUS}"="NOANSWER" & "${CFUEXT}"!=""]?docfu,1())
exten => s,n(calldocfb),GosubIf($["${PR_DIALSTATUS}"="BUSY" & "${CFBEXT}"!=""]?docfb,1())
exten => s,n,Set(DIALSTATUS=${PR_DIALSTATUS})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => docfu,1,GotoIf($["${DB(AMPUSER/${CFUEXT}/device)}" = "" ]?chlocal)
exten => docfu,n,Dial(Local/${CFUEXT}@ext-local,${RT},${DIAL_OPTIONS})
exten => docfu,n,Return()
exten => docfu,n(chlocal),ExecIf($["${DIRECTION}" = "INBOUND"]?Set(DIAL_OPTIONS=${REPLACE(DIAL_OPTIONS,T)}I))
exten => docfu,n,Dial(Local/${CFUEXT}@from-internal/n,${RT},${DIAL_OPTIONS})
exten => docfu,n,Return()
exten => docfb,1,GotoIf($["${DB(AMPUSER/${CFBEXT}/device)}" = "" ]?chlocal)
exten => docfb,n,Dial(Local/${CFBEXT}@ext-local,${RT},${DIAL_OPTIONS})
exten => docfb,n,Return()
exten => docfb,n(chlocal),ExecIf($["${DIRECTION}" = "INBOUND"]?Set(DIAL_OPTIONS=${REPLACE(DIAL_OPTIONS,T)}I))
exten => docfb,n,Dial(Local/${CFBEXT}@from-internal/n,${RT},${DIAL_OPTIONS})
exten => docfb,n,Return()
exten => _s-.,1,Noop(Extension is reporting ${EXTEN})
;--== end of [macro-simple-dial] ==--;
[macro-blkvm-setifempty]
exten => s,1,GotoIf($[!${EXISTS(${BLKVM_CHANNEL})}]?init)
exten => s,n,Set(GOSUB_RETVAL=${SHARED(BLKVM,${BLKVM_CHANNEL})})
exten => s,n,MacroExit()
exten => s,n(init),Set(__BLKVM_CHANNEL=${CHANNEL})
exten => s,n,Set(SHARED(BLKVM,${BLKVM_CHANNEL})=TRUE)
exten => s,n,Set(GOSUB_RETVAL=TRUE)
exten => s,n,MacroExit()
;--== end of [macro-blkvm-setifempty] ==--;
[macro-blkvm-set]
exten => s,1,ExecIf($[!${EXISTS(${BLKVM_CHANNEL})} | "{ARG1}" = "reset"]?Set(__BLKVM_CHANNEL=${CHANNEL}))
exten => s,n,Set(SHARED(BLKVM,${BLKVM_CHANNEL})=TRUE)
exten => s,n,Set(GOSUB_RETVAL=TRUE)
exten => s,n,MacroExit()
;--== end of [macro-blkvm-set] ==--;
[macro-blkvm-clr]
exten => s,1,Set(SHARED(BLKVM,${BLKVM_CHANNEL})=)
exten => s,n,Set(GOSUB_RETVAL=)
exten => s,n,MacroExit()
;--== end of [macro-blkvm-clr] ==--;
[macro-blkvm-check]
exten => s,1,Set(GOSUB_RETVAL=${SHARED(BLKVM,${BLKVM_CHANNEL})})
exten => s,n,ExecIf($["${GOSUB_RETVAL}"="" & "${CC_RECALL}"="1"]?Set(GOSUB_RETVAL=TRUE))
exten => s,n,MacroExit()
;--== end of [macro-blkvm-check] ==--;
[macro-hangupcall]
exten => s,1(start),GotoIf($["${USE_CONFIRMATION}"="" | "${RINGGROUP_INDEX}"="" | "${CHANNEL}"!="${UNIQCHAN}"]?theend)
exten => s,n(delrgi),Noop(Deleting: RG/${RINGGROUP_INDEX}/${CHANNEL} ${DB_DELETE(RG/${RINGGROUP_INDEX}/${CHANNEL})})
exten => s,n(theend),Hangup
exten => s,n,MacroExit()
;--== end of [macro-hangupcall] ==--;
[macro-dial-one]
exten => s,1,Set(DEXTEN=${ARG3})
exten => s,n,Set(DIALSTATUS_CW=)
exten => s,n,GosubIf($["${FROM_DID}"!="" & "${SCREEN}"="" & "${DB(AMPUSER/${DEXTEN}/screen)}"!=""]?screen,1())
exten => s,n,GosubIf($["${DB(CF/${DEXTEN})}"!=""]?cf,1())
exten => s,n,GotoIf($["${DEXTEN:-1}"="#" | "${DB(DND/${DEXTEN})}"=""]?skip1)
exten => s,n,Set(DEXTEN=)
exten => s,n,Set(DIALSTATUS=BUSY)
exten => s,n(skip1),GotoIf($["${DEXTEN}"=""]?nodial)
exten => s,n,GotoIf($["${DEXTEN:-1}"="#"]?continue)
exten => s,n,Set(EXTHASCW=${IF($["${CWIGNORE}"!=""]? :${DB(CW/${DEXTEN})})})
exten => s,n,GotoIf($["${EXTHASCW}"="" | "${DB(CFB/${DEXTEN})}"!="" | "${DB(CFU/${DEXTEN})}"!=""]?next1:cwinusebusy)
exten => s,n(next1),GotoIf($["${DB(CFU/${DEXTEN})}"!="" & ("${EXTENSION_STATE(${DEXTEN})}"="UNAVAILABLE" | "${EXTENSION_STATE(${DEXTEN})}"="UNKNOWN")]?docfu:skip3)
exten => s,n(docfu),Set(DEXTEN=)
exten => s,n,Set(DIALSTATUS=NOANSWER)
exten => s,n,Goto(nodial)
exten => s,n(skip3),GotoIf($["${EXTHASCW}"="" | "${DB(CFB/${DEXTEN})}"!=""]?next2:continue)
exten => s,n(next2),GotoIf($["${EXTENSION_STATE(${DEXTEN})}"="NOT_INUSE" | "${EXTENSION_STATE(${DEXTEN})}"="UNAVAILABLE" | "${EXTENSION_STATE(${DEXTEN})}"="UNKNOWN"]?continue)
exten => s,n,ExecIf($["${DB(CFB/${DEXTEN})}"!="" & "${CFIGNORE}"=""]?Set(DIALSTATUS=BUSY))
exten => s,n,GotoIf($["${EXTHASCW}"!="" | "${DEXTEN:-1}"="#"]?cwinusebusy)
exten => s,n,Set(DEXTEN=)
exten => s,n,Set(DIALSTATUS=BUSY)
exten => s,n,Goto(nodial)
exten => s,n(cwinusebusy),GotoIf($["${EXTHASCW}"!="" & "${CWINUSEBUSY}"="true"]?next3:continue)
exten => s,n(next3),ExecIf($["${EXTENSION_STATE(${DEXTEN})}"!="UNAVAILABLE" & "${EXTENSION_STATE(${DEXTEN})}"!="NOT_INUSE" & "${EXTENSION_STATE(${DEXTEN})}"!="UNKNOWN"]?Set(DIALSTATUS_CW=BUSY))
exten => s,n(continue),GotoIf($["${DEXTEN}"=""]?nodial)
exten => s,n,GosubIf($["${DEXTEN:-1}"!="#"]?dstring,1():dlocal,1())
exten => s,n,GotoIf($[${LEN(${DSTRING})}=0]?nodial)
exten => s,n,GotoIf($["${DEXTEN:-1}"="#"]?skiptrace)
exten => s,n,GosubIf($[${REGEX("^[\+]?[0-9]+$" ${CALLERID(number)})} = 1]?ctset,1():ctclear,1())
exten => s,n(skiptrace),Set(D_OPTIONS=${IF($["${NODEST}"!="" & ${REGEX("(M[(]auto-blkvm[)])" ${ARG2})} != 1]?${ARG2}M(auto-blkvm):${ARG2})})
exten => s,n,ExecIf($["${ALERT_INFO}"!=""]?SIPAddHeader(Alert-Info: ${ALERT_INFO}))
exten => s,n,ExecIf($["${SIPADDHEADER}"!=""]?SIPAddHeader(${SIPADDHEADER}))
exten => s,n,ExecIf($["${MOHCLASS}"!=""]?Set(CHANNEL(musicclass)=${MOHCLASS}))
exten => s,n,GosubIf($["${QUEUEWAIT}"!=""]?qwait,1())
exten => s,n,Set(__CWIGNORE=${CWIGNORE})
exten => s,n,Set(__KEEPCID=TRUE)
exten => s,n,GotoIf($["${USEGOTO}"="1"]?usegoto,1)
exten => s,n,GotoIf($["${DB(AMPUSER/${EXTTOCALL}/cidname)}" = "" || "${DB(AMPUSER/${AMPUSER}/cidname)}" = ""]?godial)
exten => s,n,Gosub(sub-presencestate-display,s,1(${EXTTOCALL}))
exten => s,n,Set(CONNECTEDLINE(name,i)=${DB(AMPUSER/${EXTTOCALL}/cidname)}${PRESENCESTATE_DISPLAY})
exten => s,n,Set(CONNECTEDLINE(num)=${EXTTOCALL})
exten => s,n,Set(D_OPTIONS=${D_OPTIONS}I)
exten => s,n(godial),Macro(dialout-one-predial-hook,)
exten => s,n,ExecIf($["${DIRECTION}" = "INBOUND"]?Set(D_OPTIONS=${REPLACE(D_OPTIONS,T)}I))
exten => s,n,Dial(${DSTRING},${ARG1},${D_OPTIONS})
exten => s,n,ExecIf($["${DIALSTATUS}"="ANSWER" & "${CALLER_DEST}"!=""]?MacroExit())
exten => s,n,ExecIf($["${DIALSTATUS_CW}"!=""]?Set(DIALSTATUS=${DIALSTATUS_CW}))
exten => s,n,GosubIf($[("${SCREEN}"!=""&("${DIALSTATUS}"="TORTURE"|"${DIALSTATUS}"="DONTCALL"))|"${DIALSTATUS}"="ANSWER"]?s-${DIALSTATUS},1())
exten => s,n,MacroExit()
exten => s,n(nodial),ExecIf($["${DIALSTATUS}" = ""]?Set(DIALSTATUS=NOANSWER))
exten => s,n,Noop(Returned from dial-one with nothing to call and DIALSTATUS: ${DIALSTATUS})
exten => s,n,MacroExit()
exten => h,1,Macro(hangupcall,)
exten => usegoto,1,Set(USEGOTO=)
exten => usegoto,n,Goto(from-internal,${DSTRING},1)
exten => screen,1,GotoIf($["${DB(AMPUSER/${DEXTEN}/screen)}"!="nomemory" | "${CALLERID(number)}"=""]?memory)
exten => screen,n,ExecIf($[${REGEX("^[0-9a-zA-Z ]+$" ${CALLERID(number)})} = 1]?System(rm -f ${ASTVARLIBDIR}/sounds/priv-callerintros/${CALLERID(number)}.*))
exten => screen,n(memory),Set(__SCREEN=${DB(AMPUSER/${DEXTEN}/screen)})
exten => screen,n,Set(__SCREEN_EXTEN=${DEXTEN})
exten => screen,n,Set(ARG2=${ARG2}p)
exten => screen,n,Return()
exten => cf,1,Set(CFAMPUSER=${IF($["${AMPUSER}"=""]?${CALLERID(number)}:${AMPUSER})})
exten => cf,n,ExecIf($["${DB(CF/${DEXTEN})}"="${CFAMPUSER}" | "${DB(CF/${DEXTEN})}"="${REALCALLERIDNUM}" | "${CUT(CUT(BLINDTRANSFER,-,1),/,1)}" = "${DB(CF/${DEXTEN})}" | "${DEXTEN}"="${DB(CF/${DEXTEN})}"]?Return())
exten => cf,n,ExecIf($["${DB(AMPUSER/${DEXTEN}/cfringtimer)}" != "0" & "${DB(AMPUSER/${DEXTEN}/cfringtimer)}" != ""]?Set(ARG1=${IF($["${DB(AMPUSER/${DEXTEN}/cfringtimer)}"="-1"]? : ${DB(AMPUSER/${DEXTEN}/cfringtimer)})}))
exten => cf,n,Set(DEXTEN=${IF($["${CFIGNORE}"=""]?"${DB(CF/${DEXTEN})}#": )})
exten => cf,n,ExecIf($["${DEXTEN}"!=""]?Return())
exten => cf,n,Set(DIALSTATUS=NOANSWER)
exten => cf,n,Return()
exten => qwait,1,ExecIf($["${SAVEDCIDNAME}" = ""]?Set(__SAVEDCIDNAME=${CALLERID(name)}))
exten => qwait,n,Set(ELAPSED=${MATH($[${EPOCH}+30-${QUEUEWAIT}]/60,int)})
exten => qwait,n,Set(CALLERID(name)=M${ELAPSED}:${SAVEDCIDNAME})
exten => qwait,n,Return()
exten => ctset,1,Set(DB(CALLTRACE/${DEXTEN})=${CALLERID(number)})
exten => ctset,n,Return()
exten => ctclear,1,Noop(Deleting: CALLTRACE/${DEXTEN} ${DB_DELETE(CALLTRACE/${DEXTEN})})
exten => ctclear,n,Return()
exten => dstring,1,Set(DSTRING=)
exten => dstring,n,Set(DEVICES=${DB(AMPUSER/${DEXTEN}/device)})
exten => dstring,n,ExecIf($["${DEVICES}"=""]?Return())
exten => dstring,n,ExecIf($["${DEVICES:0:1}"="&"]?Set(DEVICES=${DEVICES:1}))
exten => dstring,n,Set(LOOPCNT=${FIELDQTY(DEVICES,&)})
exten => dstring,n,Set(ITER=1)
exten => dstring,n(begin),Set(THISDIAL=${DB(DEVICE/${CUT(DEVICES,&,${ITER})}/dial)})
exten => dstring,n,GosubIf($["${ASTCHANDAHDI}" = "1"]?zap2dahdi,1())
exten => dstring,n,GotoIf($["${THISDIAL:0:5}"!="PJSIP"]?doset)
exten => dstring,n,Noop(Debug: Found PJSIP Destination ${THISDIAL}, updating with PJSIP_DIAL_CONTACTS)
exten => dstring,n,Set(THISDIAL=${PJSIP_DIAL_CONTACTS(${THISDIAL:6})})
exten => dstring,n,GotoIf($["${THISDIAL}"=""]?skipset)
exten => dstring,n(doset),Set(DSTRING=${DSTRING}${THISDIAL}&)
exten => dstring,n(skipset),Set(ITER=$[${ITER}+1])
exten => dstring,n,GotoIf($[${ITER}<=${LOOPCNT}]?begin)
exten => dstring,n,ExecIf($["${DSTRING:-1}"!="&"]?Return())
exten => dstring,n,Set(DSTRING=${DSTRING:0:$[${LEN(${DSTRING})}-1]})
exten => dstring,n,Return()
exten => dlocal,1,Set(DSTRING=${IF($["${ARG1}"=""]?${DEXTEN:0:${MATH(${LEN(${DEXTEN})}-1,int)}}:Local/${DEXTEN:0:${MATH(${LEN(${DEXTEN})}-1,int)}}@from-internal/n)})
exten => dlocal,n,Set(USEGOTO=${IF($["${ARG1}"=""]?1:0)})
exten => dlocal,n,Return()
exten => zap2dahdi,1,ExecIf($["${THISDIAL}" = ""]?Return())
exten => zap2dahdi,n,Set(NEWDIAL=)
exten => zap2dahdi,n,Set(LOOPCNT2=${FIELDQTY(THISDIAL,&)})
exten => zap2dahdi,n,Set(ITER2=1)
exten => zap2dahdi,n(begin2),Set(THISPART2=${CUT(THISDIAL,&,${ITER2})})
exten => zap2dahdi,n,ExecIf($["${THISPART2:0:3}" = "ZAP"]?Set(THISPART2=DAHDI${THISPART2:3}))
exten => zap2dahdi,n,Set(NEWDIAL=${NEWDIAL}${THISPART2}&)
exten => zap2dahdi,n,Set(ITER2=$[${ITER2} + 1])
exten => zap2dahdi,n,GotoIf($[${ITER2} <= ${LOOPCNT2}]?begin2)
exten => zap2dahdi,n,Set(THISDIAL=${NEWDIAL:0:$[${LEN(${NEWDIAL})}-1]})
exten => zap2dahdi,n,Return()
exten => s-TORTURE,1,Goto(app-blackhole,musiconhold,1)
exten => s-TORTURE,n,Macro(hangupcall,)
exten => s-DONTCALL,1,Answer
exten => s-DONTCALL,n,Wait(1)
exten => s-DONTCALL,n,Zapateller()
exten => s-DONTCALL,n,Playback(ss-noservice)
exten => s-DONTCALL,n,Macro(hangupcall,)
exten => s-CHANUNAVAIL,1,Macro(vm,${SCREEN_EXTEN},BUSY,${IVR_RETVM})
exten => s-CHANUNAVAIL,n,ExecIf($["${IVR_RETVM}"!="RETURN" | "${IVR_CONTEXT}"=""]?Hangup())
exten => s-CHANUNAVAIL,n,Return()
exten => s-NOANSWER,1,Macro(vm,${SCREEN_EXTEN},BUSY,${IVR_RETVM})
exten => s-NOANSWER,n,ExecIf($["${IVR_RETVM}"!="RETURN" | "${IVR_CONTEXT}"=""]?Hangup())
exten => s-NOANSWER,n,Return()
exten => s-BUSY,1,Macro(vm,${SCREEN_EXTEN},BUSY,${IVR_RETVM})
exten => s-BUSY,n,ExecIf($["${IVR_RETVM}"!="RETURN" | "${IVR_CONTEXT}"=""]?Hangup())
exten => s-BUSY,n,Return()
;--== end of [macro-dial-one] ==--;
[sipstation-welcome]
exten => _X.,1,Set(ISNUM=${REGEX("[0-9]" ${CALLERID(number)})})
exten => _X.,n,Set(DB(sipstation/${EXTEN}/lastcall/cnum)=${CALLERID(number)})
exten => _X.,n,Set(DB(sipstation/${EXTEN}/lastcall/cnam)=${CALLERID(name)})
exten => _X.,n,Set(DB(sipstation/${EXTEN}/lastcall/time)=${EPOCH})
exten => _X.,n,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,Playback(you-have-reached-a-test-number&silence/1)
exten => _X.,n,SayDigits(${EXTEN})
exten => _X.,n,Playback(your&calling&from&silence/1)
exten => _X.,n,GotoIf($["${ISNUM}" = "1"]?valid:notvalid)
exten => _X.,n(valid),SayDigits(${CALLERID(number)})
exten => _X.,n,Hangup
exten => _X.,n(notvalid),Playback(unavailable&number)
exten => _X.,n,Hangup
;--== end of [sipstation-welcome] ==--;
[from-internal-additional]
include => app-blacklist
include => app-cf-toggle
include => app-cf-unavailable-prompt-on
include => app-cf-busy-prompting-on
include => app-cf-prompting-on
include => app-cf-busy-off-any
include => app-cf-busy-off
include => app-cf-busy-on
include => app-cf-off-any
include => app-cf-off
include => app-cf-on
include => app-cf-unavailable-off
include => app-cf-unavailable-on
include => ext-cf-hints
include => app-callwaiting-cwoff
include => app-callwaiting-cwon
include => app-campon-toggle
include => app-campon-cancel
include => app-campon-request
include => ext-meetme
include => app-dictate-send
include => app-dictate-record
include => app-dnd-off
include => app-dnd-on
include => app-dnd-toggle
include => ext-dnd-hints
include => app-fax
include => app-fmf-toggle
include => ext-findmefollow
include => fmgrps
include => app-hotelwakeup
include => app-calltrace
include => app-speakextennum
include => app-speakingclock
include => app-echo-test
include => ext-intercom-users
include => park-hints
include => app-pbdirectory
include => ext-queues
include => app-recordings
include => ext-group
include => grps
include => app-speeddial
include => vmblast-grp
include => app-dialvm
include => app-vmmain
include => app-userlogonoff
include => ext-local-confirm
include => findmefollow-ringallv2
include => app-pickup
include => app-chanspy
include => ext-test
include => ext-local
include => outbound-allroutes
exten => h,1,Hangup
;--== end of [from-internal-additional] ==--;
[from-internal-noxfer-additional]
exten => h,1,Hangup
;--== end of [from-internal-noxfer-additional] ==--;pjsip.aor.conf:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip.aor_custom.conf
[300]
type=aor
mailboxes=300@device
max_contacts=1
remove_existing=yes
qualify_frequency=60
[Telekom]
type=aor
qualify_frequency=60
contact=sip:+49{Telefonnummer Intern}@tel.t-online.de:5060pjsip.auth.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip.auth_custom.conf
[300-auth]
type=auth
auth_type=userpass
password={password}
username=300
[Telekom]
type=auth
auth_type=userpass
password={password}
username=0{Telefonnummer Intern}pjsip.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip_custom.conf
#include pjsip.transports.conf
#include pjsip.transports_custom_post.conf
#include pjsip.endpoint.conf
#include pjsip.endpoint_custom_post.conf
#include pjsip.aor.conf
#include pjsip.aor_custom_post.conf
#include pjsip.auth.conf
#include pjsip.auth_custom_post.conf
#include pjsip.registration.conf
#include pjsip.registration_custom_post.conf
#include pjsip.identify.conf
#include pjsip.identify_custom_post.conf
[global]
type=global
user_agent=FPBX-AsteriskNOW-12.0.76.4(13.9.1)pjsip.endpoint.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip.endpoint_custom.conf
[300]
type=endpoint
aors=300
auth=300-auth
allow=ulaw,alaw,gsm
context=from-internal
callerid=device <300>
dtmf_mode=rfc4733
use_avpf=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
media_encryption=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
[Telekom]
type=endpoint
transport=0.0.0.0-udp
context=from-pstn
disallow=all
allow=ulaw,alaw,gsm
outbound_auth=Telekom
aors=Telekompjsip.identify.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip.identify_custom.conf
[Telekom]
type=identify
endpoint=Telekom
match=tel.t-online.depjsip.registration.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip.registration_custom.conf
[Telekom]
type=registration
transport=0.0.0.0-udp
outbound_auth=Telekom
retry_interval=60
max_retries=10
expiration=3600
auth_rejection_permanent=yes
contact_user=0{Telefonnummer Intern}
server_uri=sip:tel.t-online.de
client_uri=sip:+49{Telefonnummer Intern}@tel.t-online.de:5060pjsip.transports.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include pjsip.transports_custom.conf
[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_media_address={Extern Netz}
external_signaling_address={Extern Netz}
local_net={Localnetz}/24pjsip_notify.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
#include sip_notify_custom.conf
#include sip_notify_additional.confVor diesem Versuch hatte ich Asterisk über Debian 8 installiert und konnte da nur max 1min telefonieren, rein und raus!!!
Es muss irgendetwas an der Config sein, ich habe auch keine Ahnung was.
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Content-ID: 319197
Url: https://administrator.de/contentid/319197
Ausgedruckt am: 13.11.2024 um 09:11 Uhr
20 Kommentare
Neuester Kommentar
Die Telekom lehnt den Anruf mit Q.850-Cause 21 ab - das heißt, der Anruf wird abgewiesen.
Weshalb, geht da leider nicht draus hervor (evtl. mal SIP-Debugging für den Telekom-Peer aktivieren?).
Für mich scheint der Fehler aber in der SIP-Konfiguration zu liegen, unter der Prämisse dass du überall wirklich deine korrekte Rufnummer übermittelst. Der Benutzername (aus "pjsip.auth.conf") ist nicht deine Telefonnummer sondern dein T-Online-Zugangsname inkl. "@t-online.de"-Suffix.
Und normalerweise will die Telekom die Rufnummer bei den SIP-Registrierungen im nationalen Format, nicht in E.164 haben, wie es manchmal als "client_uri=sip:+49{Telefonnummer Intern}@tel.t-online.de:5060" auftaucht. Hier solltest du mal probieren ob die Telekom hier evtl. das nationale Rufnummernformat (so funktioniert es bei mir) haben will, also nicht mit +49 sondern 0.
Am Rande:
Deaktiviere bitte möglichst flächendeckend bei dir alle Codecs außer alaw (G.711a) und G.722.
Andere Codecs unterstützt die Telekom offiziell nicht und außerdem wäre es schwachsinnig, wenn bei dir extra eine Codec-Konvertierung stattfinden müsste.
Dein Problem mit der Asterisk unter Debian 8 könnte eventuell auch gelöst werden:
Du musst zwingend unter SIP die Session-Timers aktivieren, sonst trennt die Telekom die Verbindung.
Zudem muss der SIP-Port den du verwendest zwingend auch Verbindungen von anderen als den gerade verbundenen IP-Adressen zulassen, falls du da eine Firewall vor stehen hast. Wichtig ist dabei, dass du einen anderen Port als 5060 verwendest, wenn du durch einen Router hindurch musst, der selber VoIP beherrschen könnte - FritzBoxen oder Speedports zum Beispiel knuppern dir sonst Pakete weg, die sie nicht zuordnen können.
Weshalb, geht da leider nicht draus hervor (evtl. mal SIP-Debugging für den Telekom-Peer aktivieren?).
Für mich scheint der Fehler aber in der SIP-Konfiguration zu liegen, unter der Prämisse dass du überall wirklich deine korrekte Rufnummer übermittelst. Der Benutzername (aus "pjsip.auth.conf") ist nicht deine Telefonnummer sondern dein T-Online-Zugangsname inkl. "@t-online.de"-Suffix.
Und normalerweise will die Telekom die Rufnummer bei den SIP-Registrierungen im nationalen Format, nicht in E.164 haben, wie es manchmal als "client_uri=sip:+49{Telefonnummer Intern}@tel.t-online.de:5060" auftaucht. Hier solltest du mal probieren ob die Telekom hier evtl. das nationale Rufnummernformat (so funktioniert es bei mir) haben will, also nicht mit +49 sondern 0.
Am Rande:
Deaktiviere bitte möglichst flächendeckend bei dir alle Codecs außer alaw (G.711a) und G.722.
Andere Codecs unterstützt die Telekom offiziell nicht und außerdem wäre es schwachsinnig, wenn bei dir extra eine Codec-Konvertierung stattfinden müsste.
Dein Problem mit der Asterisk unter Debian 8 könnte eventuell auch gelöst werden:
Du musst zwingend unter SIP die Session-Timers aktivieren, sonst trennt die Telekom die Verbindung.
Zudem muss der SIP-Port den du verwendest zwingend auch Verbindungen von anderen als den gerade verbundenen IP-Adressen zulassen, falls du da eine Firewall vor stehen hast. Wichtig ist dabei, dass du einen anderen Port als 5060 verwendest, wenn du durch einen Router hindurch musst, der selber VoIP beherrschen könnte - FritzBoxen oder Speedports zum Beispiel knuppern dir sonst Pakete weg, die sie nicht zuordnen können.
Danke #LordGurke für deine schnelle Reaktion,
Die Ameldedaten habe ich wie in deinem Post geändert, er konnte keine Verbinung aufbauen.
Mit meiner vorherigen Config stellt er eine Verbindung her:
Bei den Telefonnumer habe ich alle +49 durch 0 ersetz, hat leider nicht geholfen.
PJSIP Debug beim raustelefonieren
Die Ameldedaten habe ich wie in deinem Post geändert, er konnte keine Verbinung aufbauen.
Mit meiner vorherigen Config stellt er eine Verbindung her:
[2016-10-27 08:43:32] VERBOSE[1788] res_pjsip/pjsip_configuration.c: Contact Telekom/sip:0{Telefonnumer intern}@tel.t-online.de:5060 is now Reachable. RTT: 30.964 msecBei den Telefonnumer habe ich alle +49 durch 0 ersetz, hat leider nicht geholfen.
PJSIP Debug beim raustelefonieren
<--- Received SIP request (910 bytes) from UDP:{IP-Adresse CLient}:43453 --->
INVITE sip:0{Telefonnummer Extern}@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---ffc91084846d57b2
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:0{Telefonnummer Extern}@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=6d056449
Call-ID: LEce9-SQBqE06SAWJ__9RQ..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 241
v=0
o=Z 0 0 IN IP4 {IP-Adresse CLient}
s=Z
c=IN IP4 {IP-Adresse CLient}
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<--- Transmitting SIP response (530 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---ffc91084846d57b2
Call-ID: LEce9-SQBqE06SAWJ__9RQ..
From: <sip:300@{IP-Adresse Server}>;tag=6d056449
To: <sip:0{Telefonnummer Extern}@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---ffc91084846d57b2
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1477551431/a9c34fe1db7e0e02c2c66c5ecd566380",opaque="45b9065f6d7ecf7b",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Received SIP request (955 bytes) from UDP:{IP-Adresse CLient}:43453 --->
PUBLISH sip:300@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---110b919aed99f915
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:300@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=d47c2465
Call-ID: 9iPD73PIjQ0w0RJAG4gs1g..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@{IP-Adresse Server};transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
<--- Transmitting SIP response (522 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---110b919aed99f915
Call-ID: 9iPD73PIjQ0w0RJAG4gs1g..
From: <sip:300@{IP-Adresse Server}>;tag=d47c2465
To: <sip:300@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---110b919aed99f915
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1477551431/a9c34fe1db7e0e02c2c66c5ecd566380",opaque="12c5e51060b90e67",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Received SIP request (698 bytes) from UDP:{IP-Adresse CLient}:43453 --->
SUBSCRIBE sip:300@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---71374eebacae7916
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:300@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=c4775709
Call-ID: BgHqHHbtC-tlJBIv2FAwaA..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
<--- Transmitting SIP response (524 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---71374eebacae7916
Call-ID: BgHqHHbtC-tlJBIv2FAwaA..
From: <sip:300@{IP-Adresse Server}>;tag=c4775709
To: <sip:300@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---71374eebacae7916
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1477551431/a9c34fe1db7e0e02c2c66c5ecd566380",opaque="369f9c4854418452",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Received SIP request (365 bytes) from UDP:{IP-Adresse CLient}:43453 --->
ACK sip:0{Telefonnummer Extern}@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---ffc91084846d57b2
Max-Forwards: 70
To: <sip:0{Telefonnummer Extern}@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---ffc91084846d57b2
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=6d056449
Call-ID: LEce9-SQBqE06SAWJ__9RQ..
CSeq: 1 ACK
Content-Length: 0
<--- Received SIP request (1216 bytes) from UDP:{IP-Adresse CLient}:43453 --->
INVITE sip:0{Telefonnummer Extern}@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---6fad305988c513f8
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:0{Telefonnummer Extern}@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=6d056449
Call-ID: LEce9-SQBqE06SAWJ__9RQ..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477551431/a9c34fe1db7e0e02c2c66c5ecd566380",uri="sip:0{Telefonnummer Extern}@{IP-Adresse Server};transport=UDP",response="559663947244ca78856161dec94bdef8",cnonce="47248996ea124ff49af12d96ebb0e5f4",nc=00000001,qop=auth,algorithm=md5,opaque="45b9065f6d7ecf7b"
Allow-Events: presence, kpml
Content-Length: 241
v=0
o=Z 0 0 IN IP4 {IP-Adresse CLient}
s=Z
c=IN IP4 {IP-Adresse CLient}
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<--- Received SIP request (1252 bytes) from UDP:{IP-Adresse CLient}:43453 --->
PUBLISH sip:300@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---75822dd44c9c684f
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:300@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=d47c2465
Call-ID: 9iPD73PIjQ0w0RJAG4gs1g..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477551431/a9c34fe1db7e0e02c2c66c5ecd566380",uri="sip:300@{IP-Adresse Server};transport=UDP",response="b80045db4ddb4ffe9ea7bac9ad2aac63",cnonce="640027d0f9647acfd3c05eb1386ef27a",nc=00000001,qop=auth,algorithm=md5,opaque="12c5e51060b90e67"
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@{IP-Adresse Server};transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[2016-10-27 08:57:11] WARNING[2575]: res_pjsip_pubsub.c:3008 pubsub_on_rx_publish_request: No registered publish handler for event presence
<--- Transmitting SIP response (372 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---75822dd44c9c684f
Call-ID: 9iPD73PIjQ0w0RJAG4gs1g..
From: <sip:300@{IP-Adresse Server}>;tag=d47c2465
To: <sip:300@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---75822dd44c9c684f
CSeq: 2 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Received SIP request (995 bytes) from UDP:{IP-Adresse CLient}:43453 --->
SUBSCRIBE sip:300@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---82ffc4d2815b49d2
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:300@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=c4775709
Call-ID: BgHqHHbtC-tlJBIv2FAwaA..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477551431/a9c34fe1db7e0e02c2c66c5ecd566380",uri="sip:300@{IP-Adresse Server};transport=UDP",response="834de50892693b52bc26adad7b319d42",cnonce="a0f9efbb5e55c38150279b22e7e434d6",nc=00000001,qop=auth,algorithm=md5,opaque="369f9c4854418452"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[2016-10-27 08:57:11] WARNING[2575]: res_pjsip_pubsub.c:625 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<--- Transmitting SIP response (374 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---82ffc4d2815b49d2
Call-ID: BgHqHHbtC-tlJBIv2FAwaA..
From: <sip:300@{IP-Adresse Server}>;tag=c4775709
To: <sip:300@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---82ffc4d2815b49d2
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Transmitting SIP response (337 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---6fad305988c513f8
Call-ID: LEce9-SQBqE06SAWJ__9RQ..
From: <sip:300@{IP-Adresse Server}>;tag=6d056449
To: <sip:0{Telefonnummer Extern}@{IP-Adresse Server}>
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
-- Executing [0{Telefonnummer Extern}@from-internal:1] Macro("PJSIP/300-00000006", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("PJSIP/300-00000006", "TOUCH_MONITOR=1477551431.30") in new stack
-- Executing [s@macro-user-callerid:2] Set("PJSIP/300-00000006", "AMPUSER=300") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/300-00000006", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/300-00000006", "1?Set(REALCALLERIDNUM=300)") in new stack
-- Executing [s@macro-user-callerid:5] Set("PJSIP/300-00000006", "AMPUSER=300") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/300-00000006", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("PJSIP/300-00000006", "AMPUSERCIDNAME={IP-Adresse CLient}") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("PJSIP/300-00000006", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("PJSIP/300-00000006", "AMPUSERCID=300") in new stack
-- Executing [s@macro-user-callerid:10] Set("PJSIP/300-00000006", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("PJSIP/300-00000006", "CALLERID(all)="{IP-Adresse CLient}" <300>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("PJSIP/300-00000006", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/300-00000006", "1?Set(GROUP(concurrency_limit)=300)") in new stack
-- Executing [s@macro-user-callerid:14] GosubIf("PJSIP/300-00000006", "7?sub-ccss,s,1(from-internal,)") in new stack
-- Executing [s@sub-ccss:1] ExecIf("PJSIP/300-00000006", "0?Return()") in new stack
-- Executing [s@sub-ccss:2] Set("PJSIP/300-00000006", "CCSS_SETUP=TRUE") in new stack
-- Executing [s@sub-ccss:3] GosubIf("PJSIP/300-00000006", "0?monitor_config,1(from-internal,):monitor_default,1(from-internal,)") in new stack
-- Executing [monitor_default@sub-ccss:1] GotoIf("PJSIP/300-00000006", "0?is_exten") in new stack
-- Executing [monitor_default@sub-ccss:2] StackPop("PJSIP/300-00000006", "") in new stack
-- Executing [monitor_default@sub-ccss:3] Return("PJSIP/300-00000006", "FALSE") in new stack
-- Executing [s@macro-user-callerid:15] ExecIf("PJSIP/300-00000006", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("PJSIP/300-00000006", "1?continue") in new stack
-- Goto (macro-user-callerid,s,30)
-- Executing [s@macro-user-callerid:30] Set("PJSIP/300-00000006", "CALLERID(number)=300") in new stack
-- Executing [s@macro-user-callerid:31] Set("PJSIP/300-00000006", "CALLERID(name)={IP-Adresse CLient}") in new stack
-- Executing [s@macro-user-callerid:32] Set("PJSIP/300-00000006", "CDR(cnum)=300") in new stack
-- Executing [s@macro-user-callerid:33] Set("PJSIP/300-00000006", "CDR(cnam)={IP-Adresse CLient}") in new stack
-- Executing [s@macro-user-callerid:34] Set("PJSIP/300-00000006", "CHANNEL(language)=en") in new stack
-- Executing [0{Telefonnummer Extern}@from-internal:2] Gosub("PJSIP/300-00000006", "sub-record-check,s,1(out,0{Telefonnummer Extern},dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("PJSIP/300-00000006", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("PJSIP/300-00000006", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("PJSIP/300-00000006", "NOW=1477551431") in new stack
-- Executing [s@sub-record-check:4] Set("PJSIP/300-00000006", "__DAY=27") in new stack
-- Executing [s@sub-record-check:5] Set("PJSIP/300-00000006", "__MONTH=10") in new stack
-- Executing [s@sub-record-check:6] Set("PJSIP/300-00000006", "__YEAR=2016") in new stack
-- Executing [s@sub-record-check:7] Set("PJSIP/300-00000006", "__TIMESTR=20161027-085711") in new stack
-- Executing [s@sub-record-check:8] Set("PJSIP/300-00000006", "__FROMEXTEN=300") in new stack
-- Executing [s@sub-record-check:9] Set("PJSIP/300-00000006", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("PJSIP/300-00000006", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("PJSIP/300-00000006", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("PJSIP/300-00000006", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("PJSIP/300-00000006", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("PJSIP/300-00000006", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("PJSIP/300-00000006", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("PJSIP/300-00000006", "Outbound Recording Check from 300 to 0{Telefonnummer Extern}") in new stack
-- Executing [out@sub-record-check:2] Set("PJSIP/300-00000006", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("PJSIP/300-00000006", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("PJSIP/300-00000006", "recordcheck,1(dontcare,out,0{Telefonnummer Extern})") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/300-00000006", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/300-00000006", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("PJSIP/300-00000006", "") in new stack
-- Executing [out@sub-record-check:8] Return("PJSIP/300-00000006", "") in new stack
-- Executing [0{Telefonnummer Extern}@from-internal:3] ExecIf("PJSIP/300-00000006", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [0{Telefonnummer Extern}@from-internal:4] Set("PJSIP/300-00000006", "MOHCLASS=default") in new stack
-- Executing [0{Telefonnummer Extern}@from-internal:5] ExecIf("PJSIP/300-00000006", "1?Set(TRUNKCIDOVERRIDE="{IP-Adresse CLient}" <0{Telefonnummer Intern}>)") in new stack
-- Executing [0{Telefonnummer Extern}@from-internal:6] Set("PJSIP/300-00000006", "_NODEST=") in new stack
-- Executing [0{Telefonnummer Extern}@from-internal:7] Macro("PJSIP/300-00000006", "dialout-trunk,2,0{Telefonnummer Extern},,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("PJSIP/300-00000006", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("PJSIP/300-00000006", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("PJSIP/300-00000006", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("PJSIP/300-00000006", "DIAL_NUMBER=0{Telefonnummer Extern}") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("PJSIP/300-00000006", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("PJSIP/300-00000006", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("PJSIP/300-00000006", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("PJSIP/300-00000006", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("PJSIP/300-00000006", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("PJSIP/300-00000006", "DIAL_TRUNK_OPTIONS=Tt") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("PJSIP/300-00000006", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("PJSIP/300-00000006", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("PJSIP/300-00000006", "0?Set(REALCALLERIDNUM=300)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("PJSIP/300-00000006", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("PJSIP/300-00000006", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("PJSIP/300-00000006", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("PJSIP/300-00000006", "TRUNKOUTCID=0{Telefonnummer Intern}") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("PJSIP/300-00000006", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,14)
-- Executing [s@macro-outbound-callerid:14] ExecIf("PJSIP/300-00000006", "1?Set(CALLERID(all)=0{Telefonnummer Intern})") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("PJSIP/300-00000006", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("PJSIP/300-00000006", "1?Set(CALLERID(all)={IP-Adresse CLient} <0{Telefonnummer Intern}>)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("PJSIP/300-00000006", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:18] Set("PJSIP/300-00000006", "CDR(outbound_cnum)=0{Telefonnummer Intern}") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("PJSIP/300-00000006", "CDR(outbound_cnam)={IP-Adresse CLient}") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("PJSIP/300-00000006", "0?sub-flp-2,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("PJSIP/300-00000006", "OUTNUM=0{Telefonnummer Extern}") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("PJSIP/300-00000006", "custom=PJSIP") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("PJSIP/300-00000006", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("PJSIP/300-00000006", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("PJSIP/300-00000006", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/300-00000006", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/300-00000006", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/300-00000006", "1?Set(CONNECTEDLINE(num,i)=0{Telefonnummer Extern})") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/300-00000006", "1?Set(CONNECTEDLINE(name,i)=CID:0{Telefonnummer Intern})") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("PJSIP/300-00000006", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("PJSIP/300-00000006", "PJSIP/0{Telefonnummer Extern}@Telekom,300,Tt") in new stack
-- Called PJSIP/0{Telefonnummer Extern}@Telekom
<--- Transmitting SIP request (976 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:0{Telefonnummer Extern}@tel.t-online.de:5060 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPjCXeq2juqDXLCyoKSA9t6lij7oMTBRpsV
From: "{IP-Adresse CLient}" <sip:0{Telefonnummer Intern}@{IP-Adresse Server}>;tag=Lzrve.baS1..A3DcKWQyQVYuK5f3k5HW
To: <sip:0{Telefonnummer Extern}@tel.t-online.de>
Contact: <sip:asterisk@[IP-Adresse Nat Extern]:5060>
Call-ID: h-TH1XH..w6st.DaumK0CJSnPR-lDmdg
CSeq: 29616 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Type: application/sdp
Content-Length: 286
v=0
o=- 1527630358 1527630358 IN IP4 {IP-Adresse Server}
s=Asterisk
c=IN IP4 [IP-Adresse Nat Extern]
t=0 0
m=audio 30096 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (395 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=17765;branch=z9hG4bKPjCXeq2juqDXLCyoKSA9t6lij7oMTBRpsV
To: <sip:0{Telefonnummer Extern}@tel.t-online.de>;tag=h7g4Esbg_lrbkml1mge88vhgrjh2pei3kermjmxhb
From: "{IP-Adresse CLient}" <sip:0{Telefonnummer Intern}@{IP-Adresse Server}>;tag=Lzrve.baS1..A3DcKWQyQVYuK5f3k5HW
Call-ID: h-TH1XH..w6st.DaumK0CJSnPR-lDmdg
CSeq: 29616 INVITE
Content-Length: 0
<--- Transmitting SIP request (459 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:0{Telefonnummer Extern}@tel.t-online.de:5060 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPjCXeq2juqDXLCyoKSA9t6lij7oMTBRpsV
From: "{IP-Adresse CLient}" <sip:0{Telefonnummer Intern}@{IP-Adresse Server}>;tag=Lzrve.baS1..A3DcKWQyQVYuK5f3k5HW
To: <sip:0{Telefonnummer Extern}@tel.t-online.de>;tag=h7g4Esbg_lrbkml1mge88vhgrjh2pei3kermjmxhb
Call-ID: h-TH1XH..w6st.DaumK0CJSnPR-lDmdg
CSeq: 29616 ACK
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:23] NoOp("PJSIP/300-00000006", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("PJSIP/300-00000006", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("PJSIP/300-00000006", "RC=21") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("PJSIP/300-00000006", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("PJSIP/300-00000006", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("PJSIP/300-00000006", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("PJSIP/300-00000006", "CALLERID(number)=300") in new stack
-- Executing [0{Telefonnummer Extern}@from-internal:8] Macro("PJSIP/300-00000006", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("PJSIP/300-00000006", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("PJSIP/300-00000006", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("PJSIP/300-00000006", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("PJSIP/300-00000006", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
<--- Transmitting SIP response (909 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---6fad305988c513f8
Call-ID: LEce9-SQBqE06SAWJ__9RQ..
From: <sip:300@{IP-Adresse Server}>;tag=6d056449
To: <sip:0{Telefonnummer Extern}@{IP-Adresse Server}>;tag=T3KgJ1UVI2DcfviQbZdxUdhd..5ves-h
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Contact: <sip:{IP-Adresse Server}:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
P-Asserted-Identity: "CID:0{Telefonnummer Intern}" <sip:0{Telefonnummer Extern}@{IP-Adresse Server}>
Content-Type: application/sdp
Content-Length: 270
v=0
o=- 0 2 IN IP4 {IP-Adresse Server}
s=Asterisk
c=IN IP4 {IP-Adresse Server}
t=0 0
m=audio 30082 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
-- <PJSIP/300-00000006> Playing 'all-circuits-busy-now.ulaw' (language 'en')
-- <PJSIP/300-00000006> Playing 'pls-try-call-later.ulaw' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("PJSIP/300-00000006", "20") in new stack
<--- Transmitting SIP response (596 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---6fad305988c513f8
Call-ID: LEce9-SQBqE06SAWJ__9RQ..
From: <sip:300@{IP-Adresse Server}>;tag=6d056449
To: <sip:0{Telefonnummer Extern}@{IP-Adresse Server}>;tag=T3KgJ1UVI2DcfviQbZdxUdhd..5ves-h
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Reason: Q.850;cause=21
P-Asserted-Identity: "CID:0{Telefonnummer Intern}" <sip:0{Telefonnummer Extern}@{IP-Adresse Server}>
Content-Length: 0
[2016-10-27 08:57:16] WARNING[2585][C-00000003]: channel.c:4883 ast_prod: Prodding channel 'PJSIP/300-00000006' failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'PJSIP/300-00000006' in macro 'outisbusy'
== Spawn extension (from-internal, 0{Telefonnummer Extern}, 8) exited non-zero on 'PJSIP/300-00000006'
-- Executing [h@from-internal:1] Hangup("PJSIP/300-00000006", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/300-00000006'
<--- Received SIP request (362 bytes) from UDP:{IP-Adresse CLient}:43453 --->
ACK sip:0{Telefonnummer Extern}@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---6fad305988c513f8
Max-Forwards: 70
To: <sip:0{Telefonnummer Extern}@{IP-Adresse Server}>;tag=T3KgJ1UVI2DcfviQbZdxUdhd..5ves-h
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=6d056449
Call-ID: LEce9-SQBqE06SAWJ__9RQ..
CSeq: 2 ACK
Content-Length: 0
<--- Transmitting SIP request (482 bytes) to UDP:{IP-Adresse CLient}:43453 --->
OPTIONS sip:300@{IP-Adresse CLient}:43453;rinstance=f3c621428cfef1d9 SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse Server}:5060;rport;branch=z9hG4bKPjTmdYn1A8mrMwRHi7gM.uc5i.7eH-7XiD
From: <sip:300@{IP-Adresse Server}>;tag=T7h4g-KpfwZegkYtoGXwvuVxNuBbJ-Fm
To: <sip:300@{IP-Adresse CLient};rinstance=f3c621428cfef1d9>
Contact: <sip:300@{IP-Adresse Server}:5060>
Call-ID: wW47iYUqP.W8pldxAr6UV3mh.sYSziNL
CSeq: 58514 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Received SIP response (673 bytes) from UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP {IP-Adresse Server}:5060;rport=5060;branch=z9hG4bKPjTmdYn1A8mrMwRHi7gM.uc5i.7eH-7XiD
Contact: <sip:{IP-Adresse CLient}:43453>
To: <sip:300@{IP-Adresse CLient};rinstance=f3c621428cfef1d9>;tag=023b4c55
From: <sip:300@{IP-Adresse Server}>;tag=T7h4g-KpfwZegkYtoGXwvuVxNuBbJ-Fm
Call-ID: wW47iYUqP.W8pldxAr6UV3mh.sYSziNL
CSeq: 58514 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 0
<--- Received SIP request (949 bytes) from UDP:{IP-Adresse CLient}:43453 --->
PUBLISH sip:300@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---ab009f1f8ca4cc48
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:300@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=8a1d8c57
Call-ID: tUAuE2XOWkpDNG_A-I009w..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 260
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@{IP-Adresse Server};transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>
<--- Transmitting SIP response (522 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---ab009f1f8ca4cc48
Call-ID: tUAuE2XOWkpDNG_A-I009w..
From: <sip:300@{IP-Adresse Server}>;tag=8a1d8c57
To: <sip:300@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---ab009f1f8ca4cc48
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1477551438/6c7a6c6ee8b84dd8c0e2336b72530832",opaque="6bef035560695a49",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Received SIP request (698 bytes) from UDP:{IP-Adresse CLient}:43453 --->
SUBSCRIBE sip:300@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---8560c01aa7acb74d
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:300@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=0b752514
Call-ID: ICkXmfLfQRpqGyCUOYaxfg..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
<--- Transmitting SIP response (524 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---8560c01aa7acb74d
Call-ID: ICkXmfLfQRpqGyCUOYaxfg..
From: <sip:300@{IP-Adresse Server}>;tag=0b752514
To: <sip:300@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---8560c01aa7acb74d
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1477551438/6c7a6c6ee8b84dd8c0e2336b72530832",opaque="5cc9fe265bbafbc8",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Received SIP request (1246 bytes) from UDP:{IP-Adresse CLient}:43453 --->
PUBLISH sip:300@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---1f380c2d123fa817
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:300@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=8a1d8c57
Call-ID: tUAuE2XOWkpDNG_A-I009w..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477551438/6c7a6c6ee8b84dd8c0e2336b72530832",uri="sip:300@{IP-Adresse Server};transport=UDP",response="23e73f4298dc8d8dc35b5e8054214b3f",cnonce="c1cb55c770c75db9a1a9dcaa6becd624",nc=00000001,qop=auth,algorithm=md5,opaque="6bef035560695a49"
Event: presence
Allow-Events: presence, kpml
Content-Length: 260
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@{IP-Adresse Server};transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>
[2016-10-27 08:57:18] WARNING[1783]: res_pjsip_pubsub.c:3008 pubsub_on_rx_publish_request: No registered publish handler for event presence
<--- Transmitting SIP response (372 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---1f380c2d123fa817
Call-ID: tUAuE2XOWkpDNG_A-I009w..
From: <sip:300@{IP-Adresse Server}>;tag=8a1d8c57
To: <sip:300@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---1f380c2d123fa817
CSeq: 2 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
<--- Received SIP request (995 bytes) from UDP:{IP-Adresse CLient}:43453 --->
SUBSCRIBE sip:300@{IP-Adresse Server};transport=UDP SIP/2.0
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;branch=z9hG4bK-524287-1---26154e46603fcf01
Max-Forwards: 70
Contact: <sip:300@{IP-Adresse CLient}:43453;transport=UDP>
To: <sip:300@{IP-Adresse Server};transport=UDP>
From: <sip:300@{IP-Adresse Server};transport=UDP>;tag=0b752514
Call-ID: ICkXmfLfQRpqGyCUOYaxfg..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477551438/6c7a6c6ee8b84dd8c0e2336b72530832",uri="sip:300@{IP-Adresse Server};transport=UDP",response="d8dac1f96d6d88101b29bc62640c9197",cnonce="5e1459aef3577302cbaf0a6bf27fd181",nc=00000001,qop=auth,algorithm=md5,opaque="5cc9fe265bbafbc8"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[2016-10-27 08:57:18] WARNING[1783]: res_pjsip_pubsub.c:625 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<--- Transmitting SIP response (374 bytes) to UDP:{IP-Adresse CLient}:43453 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP {IP-Adresse CLient}:43453;rport=43453;received={IP-Adresse CLient};branch=z9hG4bK-524287-1---26154e46603fcf01
Call-ID: ICkXmfLfQRpqGyCUOYaxfg..
From: <sip:300@{IP-Adresse Server}>;tag=0b752514
To: <sip:300@{IP-Adresse Server}>;tag=z9hG4bK-524287-1---26154e46603fcf01
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
Rein telefonieren kann kann ich.
Nur mal OT weil das hier immer wieder die Frage danach triggert...Dazu wäre mal deine inbound Firewall Regel auf dem WAN Port spannend zu wissen. Das könnte man dann hier mal posten damit andere das nicht immer und immer wieder nachfragen müssen weil sie daran scheitern SIP und RTP auf ein inbound VoIP Device zu forwarden.
Ok, danke für den Hinweis.
Habe oben die Inbound Firewall Regel angehängt
Habe oben die Inbound Firewall Regel angehängt
Du hast scheinbar Problem in der Bedienung der Bilder Upload Funktion, denn dort steht leider nur der URL nicht aber das embeddete Bild 
- Kamerasymbol links klicken
- Bild auswählen und hochladen
- Schreibmarke an die Position klicken wo das Bild hin soll
- + klicken.
- Im Text steht dann sowas wie image c09c702a3127d... in eckigen Klammern und NICHT der http URL !
Danke, habe ich geändert
Perfekt ! Danke !!
Wenn da als CallerID an die Telekom tatsächlich - egal ob im Name- oder Number-Teil etwas anderes als deine Telefonnummer gesendet wird (wie hier die interne IP-Adresse deines Clients) kann ich gut verstehen warum die Telekom das ablehnt.
Ok, und was kann ich dagegen unternehmen? Welche Einstellungen müsste ich anpassen?
Und vielen Dank für deine Hilfe.
Und vielen Dank für deine Hilfe.
Ich habe durch deinen Tipp jetzt eine Lösung gefunden.
Ich muss die pjsip.endpoint.conf im Unterpunkt [Telekom] manuel um diese Punkte erweitern:
Weiß du/jemand wie ich diese Änderung über die "pjsip.endpoint_custom.conf" vornehmen kann?
Die manuel geänderten Punkte werden ja bei jeder Speicherung übers Webinterface gelöscht.
Und weiß jemand zufällig wie ich über FreePBX LDAP nutze?
Ich muss die pjsip.endpoint.conf im Unterpunkt [Telekom] manuel um diese Punkte erweitern:
callerid=0[Telefonnummer]
from_user=0[Telefonnummer]
from_domain=tel.t-online.deWeiß du/jemand wie ich diese Änderung über die "pjsip.endpoint_custom.conf" vornehmen kann?
Die manuel geänderten Punkte werden ja bei jeder Speicherung übers Webinterface gelöscht.
Und weiß jemand zufällig wie ich über FreePBX LDAP nutze?
Ich habe leider keine Ahnung von FreePBX, ich administriere meine Asterisken alle von Hand...
Gibt es bei FreePBX eventuell in den Einstellungen ein Menü für benutzerdefinierte Konfigurationen?
Gibt es bei FreePBX eventuell in den Einstellungen ein Menü für benutzerdefinierte Konfigurationen?
Ich habe keine gefunde.
Um das Problem zu umgehen, habe ich es über das pure Asterisk versucht, da bekomme ich aber keine funktionierende Config hin. Ich werde das erstmal so lassen.
Euch allen danke für euren Mühen, besondern Dank an #LordGurke.
Um das Problem zu umgehen, habe ich es über das pure Asterisk versucht, da bekomme ich aber keine funktionierende Config hin. Ich werde das erstmal so lassen.
Euch allen danke für euren Mühen, besondern Dank an #LordGurke.
da bekomme ich aber keine funktionierende Config hin
Was ist der Grund ?? Wenn es beim Lord sogar mehrfach problemlos rennt wie bei Millionen anderer Asterisk User auch, kann das ja dann eigentlich keinen technischen Grund haben...?!
Ich kann dir nicht sagen woran das liegt.
Auf jedenfall habe ich Asterisk 14.11 einmal komplett neu aufgesetzt und die Config für pjsip, rtp von Asterisk Now genommen. Config in pjsip angepast, der Context wurde geändert:
extensions.conf
Jetzt habe ich noch zwei Fehler.
1. Beim reintelefonieren bekomme ich kein Ton. Auf beiden Seiten nicht.
2. Beim Raustelefonieren wird auf das SIP-Software-Telefon umgeleitet.
Full.log
Auf jedenfall habe ich Asterisk 14.11 einmal komplett neu aufgesetzt und die Config für pjsip, rtp von Asterisk Now genommen. Config in pjsip angepast, der Context wurde geändert:
extensions.conf
[general]
static=yes
writeprotect=yes
autofallthrough=yes
extenpatternmatchnew=no
clearglobalvars=no
userscontext=unspecified
icesupport=yes
[unspecified]
; wer hier landet ist entweder schlecht konfiguriert oder hat keine "Rechte"
exten => _X.,1,Answer()
exten => _X.,2,Verbose(D E F A U L T ==> ${CALLERID(num)} kam um ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} in UNSPECIFIED an als er versuchte die Nummer ${EXTEN} anzurufen.)
exten => _X.,3,Hangup()
[internalsip]
;national, mit +49 gewaehlt
exten => _+49X.,1,Dial(PJSIP/Telekom/sip:0${EXTEN:3}@tel.t-online.de,60)
exten => _+49X.,n,Hangup()
;international lassen wir nicht zu
exten => _+X.,1,Hangup()
exten => _00X.,1,Hangup()
;national, mit 0 vorneweg
exten => _0Z.,1,Dial(PJSIP/Telekom/sip:${EXTEN}@tel.t-online.de,60)
exten => _0Z.,n,Hangup()
; Ortsnetz
exten => _Z.,1,Dial(PJSIP/Telekom/sip:Ortsnetzkennzahl-mit-0${EXTEN}@tel.t-online.de,60)
exten => _Z.,n,Hangup()
[Telekom]
;exten => 0[Telefonnummer Intern],1,Noop(Processing an incoming call)
;same => n,Dial(PJSIP/user)
;same => n,Hangup()
; Anrufe von extern via Telekom
; 30 Sekunden klingen
;exten => +49[Telefonnummer Intern],1,Dial(PJSIP/user,30)
exten => 0[Telefonnummer Intern],1,Dial(PJSIP/300,15)
; danach auf die Mailbox umleiten
;exten => 0[Telefonnummer Intern],n,VoiceMail(mailboxname@mailboxcontext)
;exten => +49[Telefonnummer Intern],n,Hangup()
;;exten => 0[Telefonnummer Intern],n,Hangup()Jetzt habe ich noch zwei Fehler.
1. Beim reintelefonieren bekomme ich kein Ton. Auf beiden Seiten nicht.
2. Beim Raustelefonieren wird auf das SIP-Software-Telefon umgeleitet.
Full.log
[Oct 30 12:56:51] Asterisk 14.1.1 built by root @ asterisk on a x86_64 running Linux on 2016-10-30 10:08:53 UTC
[Oct 30 12:56:51] NOTICE[2412] dnsmgr.c: Managed DNS entries will be refreshed every 300 seconds.
[Oct 30 12:56:52] NOTICE[2412] cdr.c: CDR simple logging enabled.
[Oct 30 12:56:52] WARNING[2412] ccss.c: Could not find valid ccss.conf file. Using cc_max_requests default
[Oct 30 12:56:52] WARNING[2412] ccss.c: Could not find valid ccss.conf file. Using cc_[state]_devstate defaults
[Oct 30 12:56:52] NOTICE[2412] loader.c: 80 modules will be loaded.
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2439] res_pjsip/pjsip_resolver.c: 'UDP+IPv4' is an available SIP transport
[Oct 30 12:56:52] VERBOSE[2439] res_pjsip/pjsip_resolver.c: 'TCP+IPv4' is not an available SIP transport, disabling resolver support for it
[Oct 30 12:56:52] VERBOSE[2439] res_pjsip/pjsip_resolver.c: 'TLS+IPv4' is not an available SIP transport, disabling resolver support for it
[Oct 30 12:56:52] VERBOSE[2439] res_pjsip/pjsip_resolver.c: 'UDP+IPv6' is not an available SIP transport, disabling resolver support for it
[Oct 30 12:56:52] VERBOSE[2439] res_pjsip/pjsip_resolver.c: 'TCP+IPv6' is not an available SIP transport, disabling resolver support for it
[Oct 30 12:56:52] VERBOSE[2439] res_pjsip/pjsip_resolver.c: 'TLS+IPv6' is not an available SIP transport, disabling resolver support for it
[Oct 30 12:56:52] VERBOSE[2412] manager.c: Manager registered action PJSIPQualify
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip.so => (Basic SIP resource)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_outbound_publish.so.
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_outbound_publish.so => (PJSIP Outbound Publish Support)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_pubsub.so.
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] manager.c: Manager registered action PJSIPShowSubscriptionsInbound
[Oct 30 12:56:52] VERBOSE[2412] manager.c: Manager registered action PJSIPShowSubscriptionsOutbound
[Oct 30 12:56:52] VERBOSE[2412] manager.c: Manager registered action PJSIPShowResourceLists
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_pubsub.so => (PJSIP event resource)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_session.so.
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_session.so => (PJSIP Session resource)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_timing_timerfd.so.
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_timing_timerfd.so => (Timerfd Timing Interface)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_authenticator_digest.so.
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_authenticator_digest.so => (PJSIP authentication resource)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_endpoint_identifier_ip.so.
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_endpoint_identifier_ip.so => (PJSIP IP endpoint identifier)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_endpoint_identifier_user.so.
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_endpoint_identifier_user.so => (PJSIP username endpoint identifier)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_registrar.so.
[Oct 30 12:56:52] VERBOSE[2412] manager.c: Manager registered action PJSIPShowRegistrationsInbound
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_registrar.so => (PJSIP Registrar Support)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_dialog_info_body_generator.so.
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_dialog_info_body_generator.so => (PJSIP Extension State Dialog Info+XML Provider)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_pjsip_xpidf_body_generator.so.
[Oct 30 12:56:52] VERBOSE[2412] loader.c: res_pjsip_xpidf_body_generator.so => (PJSIP Extension State PIDF Provider)
[Oct 30 12:56:52] VERBOSE[2412] loader.c: Loading res_rtp_asterisk.so.
[Oct 30 12:56:52] VERBOSE[2412] rtp_engine.c: Registered RTP engine 'asterisk'
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/rtp.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/rtp_additional.conf': Found
[Oct 30 12:56:52] VERBOSE[2412] config.c: Parsing '/etc/asterisk/rtp_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] res_rtp_asterisk.c: RTP Allocating from port range 30000 -> 30100
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_rtp_asterisk.so => (Asterisk RTP Stack)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_musiconhold.so.
[Oct 30 12:56:53] WARNING[2412] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'MusicOnHold'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'StartMusicOnHold'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'StopMusicOnHold'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_musiconhold.so => (Music On Hold Resource)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_outbound_authenticator_digest.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_outbound_authenticator_digest.so => (PJSIP authentication resource)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_mwi_body_generator.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_mwi_body_generator.so => (PJSIP MWI resource)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_pidf_digium_body_supplement.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_pidf_digium_body_supplement.so => (PJSIP PIDF Digium presence supplement)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_pidf_eyebeam_body_supplement.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_pidf_eyebeam_body_supplement.so => (PJSIP PIDF Eyebeam supplement)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_pidf_body_generator.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_pidf_body_generator.so => (PJSIP Extension State PIDF Provider)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_exten_state.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_exten_state.so => (PJSIP Extension State Notifications)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_mwi.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_mwi.so => (PJSIP MWI resource)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_publish_asterisk.so.
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_publish_asterisk.so => (PJSIP Asterisk Event PUBLISH Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_t38.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_t38.so => (PJSIP T.38 UDPTL Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_sdp_rtp.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_sdp_rtp.so => (PJSIP SDP RTP/AVP stream handler)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading chan_pjsip.so.
[Oct 30 12:56:53] VERBOSE[2412] rtp_engine.c: Registered RTP glue 'PJSIP'
[Oct 30 12:56:53] VERBOSE[2412] channel.c: Registered channel type 'PJSIP' (PJSIP Channel Driver)
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'PJSIP_DIAL_CONTACTS'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'PJSIP_MEDIA_OFFER'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'PJSIP_SEND_SESSION_REFRESH'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: chan_pjsip.so => (PJSIP Channel Driver)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_path.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_path.so => (PJSIP Path Header Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_transport_websocket.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_send_to_voicemail.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_send_to_voicemail.so => (PJSIP REFER Send to Voicemail Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_acl.so.
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_acl.so => (PJSIP ACL Resource)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_stack.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'StackPop'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'Return'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'GosubIf'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'Gosub'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'LOCAL'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'LOCAL_PEEK'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'STACK_PEEK'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_stack.so => (Dialplan subroutines (Gosub, Return, etc))
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_multihomed.so.
[Oct 30 12:56:53] VERBOSE[2412] res_pjsip_multihomed.c: Performing DNS resolution of local hostname 'asterisk' to get local IPv4 and IPv6 address
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_multihomed.so => (PJSIP Multihomed Routing Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_caller_id.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_caller_id.so => (PJSIP Caller ID Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading format_gsm.so.
[Oct 30 12:56:53] VERBOSE[2412] file.c: Registered file format gsm, extension(s) gsm
[Oct 30 12:56:53] VERBOSE[2412] loader.c: format_gsm.so => (Raw GSM data)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_diversion.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_diversion.so => (PJSIP Add Diversion Header Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_dtmf_info.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_dtmf_info.so => (PJSIP DTMF INFO Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_header_funcs.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'PJSIP_HEADER'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_header_funcs.so => (PJSIP Header Functions)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_logger.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_logger.so => (PJSIP Packet Logger)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_messaging.so.
[Oct 30 12:56:53] VERBOSE[2412] message.c: Message technology 'pjsip' registered.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_messaging.so => (PJSIP Messaging Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading format_pcm.so.
[Oct 30 12:56:53] VERBOSE[2412] file.c: Registered file format pcm, extension(s) pcm|ulaw|ul|mu|ulw
[Oct 30 12:56:53] VERBOSE[2412] file.c: Registered file format alaw, extension(s) alaw|al|alw
[Oct 30 12:56:53] VERBOSE[2412] file.c: Registered file format au, extension(s) au
[Oct 30 12:56:53] VERBOSE[2412] file.c: Registered file format g722, extension(s) g722
[Oct 30 12:56:53] VERBOSE[2412] loader.c: format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading format_wav_gsm.so.
[Oct 30 12:56:53] VERBOSE[2412] file.c: Registered file format wav49, extension(s) WAV|wav49
[Oct 30 12:56:53] VERBOSE[2412] loader.c: format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM))
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_nat.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_nat.so => (PJSIP NAT Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_notify.so.
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_notify.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action PJSIPNotify
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_notify.so => (CLI/AMI PJSIP NOTIFY Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading format_wav.so.
[Oct 30 12:56:53] VERBOSE[2412] file.c: Registered file format wav, extension(s) wav
[Oct 30 12:56:53] VERBOSE[2412] file.c: Registered file format wav16, extension(s) wav16
[Oct 30 12:56:53] VERBOSE[2412] loader.c: format_wav.so => (Microsoft WAV/WAV16 format (8kHz/16kHz Signed Linear))
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_one_touch_record_info.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_one_touch_record_info.so => (PJSIP INFO One Touch Recording Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_outbound_registration.so.
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action PJSIPUnregister
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action PJSIPRegister
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action PJSIPShowRegistrationsOutbound
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.transports_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.endpoint_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.aor_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.auth_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.registration_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/pjsip.identify_custom_post.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_outbound_registration.so => (PJSIP Outbound Registration Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_rfc3326.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_rfc3326.so => (PJSIP RFC3326 Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_registrar_expire.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_registrar_expire.so => (PJSIP Contact Auto-Expiration)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading res_pjsip_refer.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: res_pjsip_refer.so => (PJSIP Blind and Attended Transfer Support)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_confbridge.so.
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/confbridge.conf': Found
[Oct 30 12:56:53] NOTICE[2412] confbridge/conf_config_parser.c: Adding default_bridge profile to app_confbridge
[Oct 30 12:56:53] NOTICE[2412] confbridge/conf_config_parser.c: Adding default_user profile to app_confbridge
[Oct 30 12:56:53] NOTICE[2412] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Oct 30 12:56:53] VERBOSE[2412] channel.c: Registered channel type 'CBRec' (Conference Bridge Recording Channel)
[Oct 30 12:56:53] VERBOSE[2412] channel.c: Registered channel type 'CBAnn' (Conference Bridge Announcing Channel)
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'ConfBridge'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'CONFBRIDGE'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'CONFBRIDGE_INFO'
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeList
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeListRooms
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeMute
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeUnmute
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeKick
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeUnlock
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeLock
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeStartRecord
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeStopRecord
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action ConfbridgeSetSingleVideoSrc
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_confbridge.so => (Conference Bridge Application)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading func_devstate.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'DEVICE_STATE'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'HINT'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: func_devstate.so => (Gets or sets a device state in the dialplan)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading cdr_custom.so.
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/cdr_custom.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] loader.c: cdr_custom.so => (Customizable Comma Separated Values CDR Backend)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading codec_gsm.so.
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'gsmtolin' from codec gsm to slin, table cost, 900000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'lintogsm' from codec slin to gsm, table cost, 600000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] loader.c: codec_gsm.so => (GSM Coder/Decoder)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_playback.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'Playback'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_playback.so => (Sound File Playback Application)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading bridge_softmix.so.
[Oct 30 12:56:53] VERBOSE[2412] bridge.c: Registered bridge technology softmix
[Oct 30 12:56:53] VERBOSE[2412] loader.c: bridge_softmix.so => (Multi-party software based channel mixing)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_bridgewait.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'BridgeWait'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_bridgewait.so => (Place the channel into a holding bridge application)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading codec_g722.so.
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'g722tolin' from codec g722 to slin, table cost, 960000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'lintog722' from codec slin to g722, table cost, 825000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'g722tolin16' from codec g722 to slin, table cost, 900000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'lin16tog722' from codec slin to g722, table cost, 600000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] loader.c: codec_g722.so => (ITU G.722-64kbps G722 Transcoder)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_dial.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'Dial'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'RetryDial'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_dial.so => (Dialing Application)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading func_strings.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'FIELDQTY'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'FIELDNUM'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'FILTER'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'REPLACE'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'STRREPLACE'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'LISTFILTER'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'REGEX'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'ARRAY'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUOTE'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'CSV_QUOTE'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'LEN'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'STRFTIME'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'STRPTIME'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'EVAL'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'KEYPADHASH'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'HASHKEYS'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'HASH'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'ClearHash'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'TOUPPER'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'TOLOWER'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'SHIFT'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'POP'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'PUSH'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'UNSHIFT'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'PASSTHRU'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: func_strings.so => (String handling dialplan functions)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading bridge_holding.so.
[Oct 30 12:56:53] VERBOSE[2412] bridge.c: Registered bridge technology holding_bridge
[Oct 30 12:56:53] VERBOSE[2412] loader.c: bridge_holding.so => (Holding bridge module)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_voicemail.so.
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/voicemail.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'VoiceMail'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'VoiceMailMain'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'MailboxExists'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'VMAuthenticate'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'VoiceMailPlayMsg'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'VMSayName'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'MAILBOX_EXISTS'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'VM_INFO'
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action VoicemailUsersList
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action VoicemailRefresh
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_voicemail.so => (Comedian Mail (Voicemail System))
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading func_cdr.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'CDR'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'CDR_PROP'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: func_cdr.so => (Call Detail Record (CDR) dialplan functions)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading func_sorcery.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'AST_SORCERY'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: func_sorcery.so => (Get a field from a sorcery object)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading func_callerid.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'CALLERPRES'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'CALLERID'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'CONNECTEDLINE'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'REDIRECTING'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting))
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading chan_bridge_media.so.
[Oct 30 12:56:53] VERBOSE[2412] channel.c: Registered channel type 'Announcer' (Bridge Media Announcing Channel Driver)
[Oct 30 12:56:53] VERBOSE[2412] channel.c: Registered channel type 'Recorder' (Bridge Media Recording Channel Driver)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: chan_bridge_media.so => (Bridge Media Channel Driver)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading pbx_config.so.
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action DialplanExtensionAdd
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action DialplanExtensionRemove
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/extensions.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Registered extension context 'unspecified'; registrar: pbx_config
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_X.' priority 1 to unspecified
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_X.' priority 2 to unspecified
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_X.' priority 3 to unspecified
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Registered extension context 'internalsip'; registrar: pbx_config
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_+49X.' priority 1 to internalsip
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_+49X.' priority 2 to internalsip
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_+X.' priority 1 to internalsip
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_00X.' priority 1 to internalsip
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_0Z.' priority 1 to internalsip
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_0Z.' priority 2 to internalsip
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_Z.' priority 1 to internalsip
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '_Z.' priority 2 to internalsip
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Registered extension context 'Telekom'; registrar: pbx_config
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Added extension '0[Telefonnummer Intern]' priority 1 to Telekom
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Time to scan old dialplan and merge leftovers back into the new: 0.000001 sec
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Time to restore hints and swap in new dialplan: 0.000005 sec
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Time to delete the old dialplan: 0.000000 sec
[Oct 30 12:56:53] VERBOSE[2412] pbx.c: Total time merge_contexts_delete: 0.000006 sec
[Oct 30 12:56:53] VERBOSE[2412] loader.c: pbx_config.so => (Text Extension Configuration)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading codec_resample.so.
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 8000khz -> 12000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 8000khz -> 16000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 8000khz -> 24000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 8000khz -> 32000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 8000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 8000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 8000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 8000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 12000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 12000khz -> 16000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 12000khz -> 24000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 12000khz -> 32000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 12000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 12000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 12000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 12000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 16000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 16000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 16000khz -> 24000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 16000khz -> 32000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 16000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 16000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 16000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 16000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 24000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 24000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 24000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 24000khz -> 32000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 24000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 24000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 24000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 24000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 32000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 32000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 32000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 32000khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 32000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 32000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 32000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 32000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 44100khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 44100khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 44100khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 44100khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 44100khz -> 32000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 44100khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 44100khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 44100khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 48000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 48000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 48000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 48000khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 48000khz -> 32000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 48000khz -> 44100khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 48000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 48000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 96000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 96000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 96000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 96000khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 96000khz -> 32000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 96000khz -> 44100khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 96000khz -> 48000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 96000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 192000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 192000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 192000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 192000khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 192000khz -> 32000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 192000khz -> 44100khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 192000khz -> 48000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'slin 192000khz -> 96000khz' from codec slin to slin, table cost, 850000, computational cost 999999
[Oct 30 12:56:53] VERBOSE[2412] loader.c: codec_resample.so => (SLIN Resampling Codec)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading codec_alaw.so.
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'alawtolin' from codec alaw to slin, table cost, 900000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'lintoalaw' from codec slin to alaw, table cost, 600000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] loader.c: codec_alaw.so => (A-law Coder/Decoder)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading bridge_builtin_features.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: bridge_builtin_features.so => (Built in bridging features)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading bridge_builtin_interval_features.so.
[Oct 30 12:56:53] VERBOSE[2412] loader.c: bridge_builtin_interval_features.so => (Built in bridging interval features)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_verbose.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'Log'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'Verbose'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_verbose.so => (Send verbose output)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_directory.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'Directory'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_directory.so => (Extension Directory)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading func_pjsip_endpoint.so.
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'PJSIP_ENDPOINT'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: func_pjsip_endpoint.so => (Get information about a PJSIP endpoint)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading codec_ulaw.so.
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'ulawtolin' from codec ulaw to slin, table cost, 900000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'lintoulaw' from codec slin to ulaw, table cost, 600000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'lintotestlaw' from codec slin to testlaw, table cost, 600000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] translate.c: Registered translator 'testlawtolin' from codec testlaw to slin, table cost, 900000, computational cost 1
[Oct 30 12:56:53] VERBOSE[2412] loader.c: codec_ulaw.so => (mu-Law Coder/Decoder)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading bridge_native_rtp.so.
[Oct 30 12:56:53] VERBOSE[2412] bridge.c: Registered bridge technology native_rtp
[Oct 30 12:56:53] VERBOSE[2412] loader.c: bridge_native_rtp.so => (Native RTP bridging module)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading bridge_simple.so.
[Oct 30 12:56:53] VERBOSE[2412] bridge.c: Registered bridge technology simple_bridge
[Oct 30 12:56:53] VERBOSE[2412] loader.c: bridge_simple.so => (Simple two channel bridging module)
[Oct 30 12:56:53] VERBOSE[2412] loader.c: Loading app_queue.so.
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/queuerules.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] config.c: Parsing '/etc/asterisk/queues.conf': Found
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'Queue'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'AddQueueMember'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'RemoveQueueMember'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'PauseQueueMember'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'UnpauseQueueMember'
[Oct 30 12:56:53] VERBOSE[2412] pbx_app.c: Registered application 'QueueLog'
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action Queues
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueStatus
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueSummary
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueAdd
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueRemove
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueuePause
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueLog
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueuePenalty
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueMemberRingInUse
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueRule
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueReload
[Oct 30 12:56:53] VERBOSE[2412] manager.c: Manager registered action QueueReset
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUEUE_VARIABLES'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUEUE_EXISTS'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUEUE_MEMBER'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUEUE_MEMBER_COUNT'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUEUE_MEMBER_LIST'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUEUE_GET_CHANNEL'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUEUE_WAITING_COUNT'
[Oct 30 12:56:53] VERBOSE[2412] pbx_functions.c: Registered custom function 'QUEUE_MEMBER_PENALTY'
[Oct 30 12:56:53] VERBOSE[2412] loader.c: app_queue.so => (True Call Queueing)
[Oct 30 12:56:53] VERBOSE[2412] asterisk.c: Asterisk Ready.
[Oct 30 12:56:57] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (591 bytes) to UDP:217.0.23.100:5060 --->
REGISTER sip:tel.t-online.de:5060 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj8a642fff-0561-4753-ba4a-8fe1a42558de
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=917c4dbc-f0d4-4c8b-ac18-43cbb63750c1
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
Call-ID: 6814185d-1a81-4644-bbb3-d8b76754cbbd
CSeq: 8359 REGISTER
Contact: <sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060>
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:56:57] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (606 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 401 Unauthorized 010330345
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj8a642fff-0561-4753-ba4a-8fe1a42558de
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_8e10c15505ea492100c521dc13c4c9f2
From: <sip:+49[Telefonnummer Intern]@tel.t-online.de>;tag=917c4dbc-f0d4-4c8b-ac18-43cbb63750c1
Call-ID: 6814185d-1a81-4644-bbb3-d8b76754cbbd
CSeq: 8359 REGISTER
Service-Route: <sip:217.0.23.100:5060;transport=udp;lr>
WWW-Authenticate: Digest realm="tel.t-online.de",nonce="385B42FC14E015580000000002374E2C",stale=true,algorithm=MD5,qop="auth"
Content-Length: 0
[Oct 30 12:56:57] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (866 bytes) to UDP:217.0.23.100:5060 --->
REGISTER sip:tel.t-online.de:5060 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj1d4b4bb8-c700-4456-803a-25dae22c14b9
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=917c4dbc-f0d4-4c8b-ac18-43cbb63750c1
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
Call-ID: 6814185d-1a81-4644-bbb3-d8b76754cbbd
CSeq: 8360 REGISTER
Contact: <sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060>
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Authorization: Digest username="0[Telefonnummer Intern]", realm="tel.t-online.de", nonce="385B42FC14E015580000000002374E2C", uri="sip:tel.t-online.de:5060", response="e2ca4926b98b6db6c0315440ad6922fa", algorithm=MD5, cnonce="884a12ca-8d0e-494e-8b72-88761466747b", qop=auth, nc=00000001
Content-Length: 0
[Oct 30 12:56:57] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (745 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj1d4b4bb8-c700-4456-803a-25dae22c14b9
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_8e10c15505e9e60c40c521dc13fbc1aa
From: <sip:+49[Telefonnummer Intern]@tel.t-online.de>;tag=917c4dbc-f0d4-4c8b-ac18-43cbb63750c1
Call-ID: 6814185d-1a81-4644-bbb3-d8b76754cbbd
CSeq: 8360 REGISTER
Contact: <sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060>;expires=660
P-Associated-Uri: <sip:+49[Telefonnummer Intern]@tel.t-online.de>
P-Associated-Uri: <tel:+49[Telefonnummer Intern]>
Service-Route: <sip:217.0.23.100:5060;transport=udp;lr>
Content-Length: 0
Authentication-Info: qop=auth,rspauth="41e6278164314a1ecfd4b6b05055acf1",cnonce="884a12ca-8d0e-494e-8b72-88761466747b",nc=00000001
[Oct 30 12:57:06] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (476 bytes) to UDP:217.0.23.100:5060 --->
OPTIONS sip:0[Telefonnummer Intern]@tel.t-online.de:5060 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj6027b07c-e0ef-4a35-88e1-c3b45f75b898
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=ac03ef81-7b3c-4755-86d6-7aa55fcc4b62
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
Contact: <sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060>
Call-ID: 308f6066-8ea6-4a4a-839e-16db59729d20
CSeq: 39814 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:06] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (477 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 In Ordnung
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj6027b07c-e0ef-4a35-88e1-c3b45f75b898
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_7nh8s3q4rqkp76b6b59qizr4ngnd5pny
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=ac03ef81-7b3c-4755-86d6-7aa55fcc4b62
Call-ID: 308f6066-8ea6-4a4a-839e-16db59729d20
CSeq: 39814 OPTIONS
Content-Length: 0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,UPDATE,REFER
[Oct 30 12:57:06] VERBOSE[2446] res_pjsip/pjsip_configuration.c: Contact Telekom/sip:0[Telefonnummer Intern]@tel.t-online.de:5060 is now Reachable. RTT: 26.494 msec
[Oct 30 12:57:06] VERBOSE[2446] res_pjsip/pjsip_configuration.c: Endpoint Telekom is now Reachable
[Oct 30 12:57:11] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (908 bytes) from UDP:[IP-Adresse-Client]:57980 --->
INVITE sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---f3b60c205f9425e1
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=3f03b043
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 241
v=0
o=Z 0 0 IN IP4 [IP-Adresse-Client]
s=Z
c=IN IP4 [IP-Adresse-Client]
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[Oct 30 12:57:11] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (529 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---f3b60c205f9425e1
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
From: <sip:300@[IP-Adresse-Server]>;tag=3f03b043
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---f3b60c205f9425e1
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1477828631/2de0b28d60bfd381c5eb2d498690e707",opaque="3826623d0d468ff3",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:11] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (955 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---e468da97f44ea695
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=fd672125
Call-ID: dpQFo9saFBfeY5ZApoJ1GQ..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[Oct 30 12:57:11] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (522 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---e468da97f44ea695
Call-ID: dpQFo9saFBfeY5ZApoJ1GQ..
From: <sip:300@[IP-Adresse-Server]>;tag=fd672125
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---e468da97f44ea695
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1477828631/2de0b28d60bfd381c5eb2d498690e707",opaque="54fd61bb20e4a259",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:11] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (698 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---940bfafc084c4c55
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=cb368811
Call-ID: PleUfuO5xFZTxzTHzFix3w..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:11] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (524 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---940bfafc084c4c55
Call-ID: PleUfuO5xFZTxzTHzFix3w..
From: <sip:300@[IP-Adresse-Server]>;tag=cb368811
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---940bfafc084c4c55
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1477828631/2de0b28d60bfd381c5eb2d498690e707",opaque="121248cf41eb4db4",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:11] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (363 bytes) from UDP:[IP-Adresse-Client]:57980 --->
ACK sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---f3b60c205f9425e1
Max-Forwards: 70
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---f3b60c205f9425e1
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=3f03b043
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
CSeq: 1 ACK
Content-Length: 0
[Oct 30 12:57:11] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1213 bytes) from UDP:[IP-Adresse-Client]:57980 --->
INVITE sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---ce17818a508f924d
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=3f03b043
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828631/2de0b28d60bfd381c5eb2d498690e707",uri="sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP",response="1422829176926f505ff6520f83c8b842",cnonce="128851d32a72be48c5620e233d2cf248",nc=00000001,qop=auth,algorithm=md5,opaque="3826623d0d468ff3"
Allow-Events: presence, kpml
Content-Length: 241
v=0
o=Z 0 0 IN IP4 [IP-Adresse-Client]
s=Z
c=IN IP4 [IP-Adresse-Client]
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[Oct 30 12:57:11] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1252 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---8afb2f6a41126b94
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=fd672125
Call-ID: dpQFo9saFBfeY5ZApoJ1GQ..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828631/2de0b28d60bfd381c5eb2d498690e707",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="6309a814b90ecd75423afa1d3ab04793",cnonce="55873a9b5d043fdadae302adc0f30f51",nc=00000001,qop=auth,algorithm=md5,opaque="54fd61bb20e4a259"
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[Oct 30 12:57:11] WARNING[2440] res_pjsip_pubsub.c: No registered publish handler for event presence
[Oct 30 12:57:11] VERBOSE[2440] res_pjsip_logger.c: <--- Transmitting SIP response (372 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---8afb2f6a41126b94
Call-ID: dpQFo9saFBfeY5ZApoJ1GQ..
From: <sip:300@[IP-Adresse-Server]>;tag=fd672125
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---8afb2f6a41126b94
CSeq: 2 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:11] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (995 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---49d89f2b8375bd67
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=cb368811
Call-ID: PleUfuO5xFZTxzTHzFix3w..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828631/2de0b28d60bfd381c5eb2d498690e707",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="32e022de35008d97be281b8235fe2f07",cnonce="c176e8bb602c370193c292aad4e0ab1e",nc=00000001,qop=auth,algorithm=md5,opaque="121248cf41eb4db4"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:11] WARNING[2440] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo
[Oct 30 12:57:11] VERBOSE[2440] res_pjsip_logger.c: <--- Transmitting SIP response (374 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---49d89f2b8375bd67
Call-ID: PleUfuO5xFZTxzTHzFix3w..
From: <sip:300@[IP-Adresse-Server]>;tag=cb368811
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---49d89f2b8375bd67
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:11] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (336 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---ce17818a508f924d
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
From: <sip:300@[IP-Adresse-Server]>;tag=3f03b043
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:11] VERBOSE[2460][C-00000001] pbx.c: Executing [0[Telefonnummer Intern]@internalsip:1] Dial("PJSIP/300-00000000", "PJSIP/Telekom/sip:0[Telefonnummer Intern]@tel.t-online.de,60") in new stack
[Oct 30 12:57:11] VERBOSE[2460][C-00000001] app_dial.c: Called PJSIP/Telekom/sip:0[Telefonnummer Intern]@tel.t-online.de
[Oct 30 12:57:12] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (1007 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:0[Telefonnummer Intern]@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPja2a2b86e-a3cd-42cd-a219-6b6241a48f83
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
Contact: <sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060>
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20190 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Type: application/sdp
Content-Length: 310
v=0
o=- 579367074 579367074 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse Nat Extern]
t=0 0
m=audio 30046 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:12] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (563 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 407 Proxy Authentication Required 02035034C
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPja2a2b86e-a3cd-42cd-a219-6b6241a48f83
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_0c4634c705ea5bd2a0c521dc4e7bb194
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20190 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="2E8E950223E01558000000008B75081C",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true
[Oct 30 12:57:12] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (461 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:0[Telefonnummer Intern]@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPja2a2b86e-a3cd-42cd-a219-6b6241a48f83
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_0c4634c705ea5bd2a0c521dc4e7bb194
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20190 ACK
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:12] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (1295 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:0[Telefonnummer Intern]@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj92b8f8ba-0f31-4126-93fa-8264aed9aa12
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
Contact: <sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060>
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20191 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Proxy-Authorization: Digest username="0[Telefonnummer Intern]", realm="tel.t-online.de", nonce="2E8E950223E01558000000008B75081C", uri="sip:0[Telefonnummer Intern]@tel.t-online.de", response="520b380225d3bf05c0cfad2d2f5e5be7", algorithm=MD5, cnonce="01b53e24-a859-4ad3-b3a5-c13872489052", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 310
v=0
o=- 579367074 579367074 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse Nat Extern]
t=0 0
m=audio 30046 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:12] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (355 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj92b8f8ba-0f31-4126-93fa-8264aed9aa12
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20191 INVITE
Content-Length: 0
[Oct 30 12:57:12] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1397 bytes) from UDP:217.0.23.100:5060 --->
INVITE sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 62
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iwcchp4risvuqxuiqk8txv62pz
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
From: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s3_3400818628-2037379828
Call-ID: p65563t1477828632m832251c566091993s4
CSeq: 1 INVITE
Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Min-Se: 900
P-Asserted-Identity: <sip:+49[Telefonnummer Intern]@tel.t-online.de>
P-Asserted-Identity: <tel:+49[Telefonnummer Intern]>
Session-Expires: 1800
Supported: timer
Supported: 100rel
Supported: replaces
Supported: norefersub
User-Agent: FPBX-AsteriskNOW-12.0.76.4 (13.9.1)
Content-Type: application/sdp
Content-Length: 306
Session-ID: 572aac693a03be68bd2fe5237a3143fb
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PUBLISH, UPDATE, PRACK, INVITE, ACK, OPTIONS, CANCEL, BYE
v=0
o=- 1290627170 3400818371 IN IP4 217.0.23.100
s=Asterisk
c=IN IP4 217.0.5.71
t=0 0
m=audio 23672 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:12] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (482 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7iwcchp4risvuqxuiqk8txv62pz
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65563t1477828632m832251c566091993s4
From: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s3_3400818628-2037379828
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
CSeq: 1 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:12] VERBOSE[2461][C-00000002] pbx.c: Executing [0[Telefonnummer Intern]@Telekom:1] Dial("PJSIP/Telekom-00000002", "PJSIP/300,15") in new stack
[Oct 30 12:57:12] VERBOSE[2461][C-00000002] app_dial.c: Called PJSIP/300
[Oct 30 12:57:12] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (1098 bytes) to UDP:[IP-Adresse-Client]:57980 --->
INVITE sip:300@[IP-Adresse-Client]:57980;rinstance=ac8a4a8408b08df5 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPje86ea49d-84af-41a6-9f3f-da539c80179d
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=5782b220-e83c-41fc-986b-87dfbd79e6f9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>
Contact: <sip:asterisk@[IP-Adresse-Server]:5060>
Call-ID: 551b9c52-864e-4da8-acb8-77e0afe30143
CSeq: 15562 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Type: application/sdp
Content-Length: 310
v=0
o=- 968824427 968824427 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse-Server]
t=0 0
m=audio 30028 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:12] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (347 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPje86ea49d-84af-41a6-9f3f-da539c80179d
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=5782b220-e83c-41fc-986b-87dfbd79e6f9
Call-ID: 551b9c52-864e-4da8-acb8-77e0afe30143
CSeq: 15562 INVITE
Content-Length: 0
[Oct 30 12:57:12] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (433 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPje86ea49d-84af-41a6-9f3f-da539c80179d
Contact: <sip:300@[IP-Adresse-Client]:57980>
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=c407d96c
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=5782b220-e83c-41fc-986b-87dfbd79e6f9
Call-ID: 551b9c52-864e-4da8-acb8-77e0afe30143
CSeq: 15562 INVITE
User-Agent: Z 3.9.32144 r32121
Content-Length: 0
[Oct 30 12:57:12] VERBOSE[2461][C-00000002] app_dial.c: PJSIP/300-00000003 is ringing
[Oct 30 12:57:12] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (671 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7iwcchp4risvuqxuiqk8txv62pz
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65563t1477828632m832251c566091993s4
From: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s3_3400818628-2037379828
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=63914302-b055-4d16-99f3-bc1284a13020
CSeq: 1 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Contact: <sip:[IP-Adresse Nat Extern]:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Content-Length: 0
[Oct 30 12:57:12] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (628 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj92b8f8ba-0f31-4126-93fa-8264aed9aa12
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s1_3400723970-1344858605
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20191 INVITE
Contact: <sip:sgc_c@217.0.23.100;transport=udp>
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Supported: timer
Content-Length: 0
Allow: NOTIFY, UPDATE, PRACK, REFER, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER
[Oct 30 12:57:12] VERBOSE[2460][C-00000001] app_dial.c: PJSIP/Telekom-00000001 is ringing
[Oct 30 12:57:12] VERBOSE[2440] res_pjsip_logger.c: <--- Transmitting SIP response (580 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---ce17818a508f924d
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
From: <sip:300@[IP-Adresse-Server]>;tag=3f03b043
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=07c0486a-39c9-4d99-a414-d512021fd037
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Contact: <sip:[IP-Adresse-Server]:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (395 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPje86ea49d-84af-41a6-9f3f-da539c80179d
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=c407d96c
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=5782b220-e83c-41fc-986b-87dfbd79e6f9
Call-ID: 551b9c52-864e-4da8-acb8-77e0afe30143
CSeq: 15562 INVITE
User-Agent: Z 3.9.32144 r32121
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (467 bytes) to UDP:[IP-Adresse-Client]:57980 --->
ACK sip:300@[IP-Adresse-Client]:57980;rinstance=ac8a4a8408b08df5 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPje86ea49d-84af-41a6-9f3f-da539c80179d
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=5782b220-e83c-41fc-986b-87dfbd79e6f9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=c407d96c
Call-ID: 551b9c52-864e-4da8-acb8-77e0afe30143
CSeq: 15562 ACK
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2461][C-00000002] app_dial.c: Everyone is busy/congested at this time (1:1/0/0)
[Oct 30 12:57:15] VERBOSE[2461][C-00000002] pbx.c: Auto fallthrough, channel 'PJSIP/Telekom-00000002' status is 'BUSY'
[Oct 30 12:57:15] VERBOSE[2440] res_pjsip_logger.c: <--- Transmitting SIP response (661 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7iwcchp4risvuqxuiqk8txv62pz
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65563t1477828632m832251c566091993s4
From: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s3_3400818628-2037379828
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=63914302-b055-4d16-99f3-bc1284a13020
CSeq: 1 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Reason: Q.850;cause=17
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (955 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---a960d1fd7fb628de
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=ff02387e
Call-ID: KMtx-Hu45Eswmq_wcpFitA..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[Oct 30 12:57:15] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (522 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---a960d1fd7fb628de
Call-ID: KMtx-Hu45Eswmq_wcpFitA..
From: <sip:300@[IP-Adresse-Server]>;tag=ff02387e
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---a960d1fd7fb628de
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1477828635/4b0065943e2110e59faa5c816e4e0ca6",opaque="22455a4c77b72871",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (698 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---f6c4cc7f41b52732
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=57257857
Call-ID: uyKy8HXq_U_R6d6yJfEqqQ..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (524 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---f6c4cc7f41b52732
Call-ID: uyKy8HXq_U_R6d6yJfEqqQ..
From: <sip:300@[IP-Adresse-Server]>;tag=57257857
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---f6c4cc7f41b52732
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1477828635/4b0065943e2110e59faa5c816e4e0ca6",opaque="4866cef710f13290",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1252 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---93979ae32edb957e
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=ff02387e
Call-ID: KMtx-Hu45Eswmq_wcpFitA..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828635/4b0065943e2110e59faa5c816e4e0ca6",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="8294bc3a0101c0729e408f37bd882b1e",cnonce="e3a94d5bbd3e0b55bbc0c62ffa3a431e",nc=00000001,qop=auth,algorithm=md5,opaque="22455a4c77b72871"
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[Oct 30 12:57:15] WARNING[2439] res_pjsip_pubsub.c: No registered publish handler for event presence
[Oct 30 12:57:15] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (372 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---93979ae32edb957e
Call-ID: KMtx-Hu45Eswmq_wcpFitA..
From: <sip:300@[IP-Adresse-Server]>;tag=ff02387e
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---93979ae32edb957e
CSeq: 2 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (995 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---2d9589b3d362bb43
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=57257857
Call-ID: uyKy8HXq_U_R6d6yJfEqqQ..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828635/4b0065943e2110e59faa5c816e4e0ca6",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="3916e4d82b99f9f176a8933bfc63b4b8",cnonce="ff738c96ba9a5f5b3751c41c1ca4f35c",nc=00000001,qop=auth,algorithm=md5,opaque="4866cef710f13290"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:15] WARNING[2439] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo
[Oct 30 12:57:15] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (374 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---2d9589b3d362bb43
Call-ID: uyKy8HXq_U_R6d6yJfEqqQ..
From: <sip:300@[IP-Adresse-Server]>;tag=57257857
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---2d9589b3d362bb43
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:15] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (471 bytes) from UDP:217.0.23.100:5060 --->
ACK sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;received=217.0.23.100;rport=5060;branch=z9hG4bKg3Zqkv7iwcchp4risvuqxuiqk8txv62pz
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=63914302-b055-4d16-99f3-bc1284a13020
From: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s3_3400818628-2037379828
Call-ID: p65563t1477828632m832251c566091993s4
CSeq: 1 ACK
Content-Length: 0
[Oct 30 12:57:16] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (961 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj92b8f8ba-0f31-4126-93fa-8264aed9aa12
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s1_3404367918-1
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20191 INVITE
Contact: <sip:sgc_c@217.0.23.100;transport=udp>
Record-Route: <sip:217.0.23.100;transport=udp;lr>
P-Early-Media: sendrecv
Require: 100rel
RSeq: 2
Supported: timer
Content-Type: application/sdp
Content-Length: 238
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, UPDATE, PRACK, INVITE, ACK, OPTIONS, CANCEL, BYE
v=0
o=- 1172654467 3404367903 IN IP4 217.0.23.100
s=media server session
t=0 0
m=audio 13692 RTP/AVP 9 101
c=IN IP4 217.0.4.231
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32,36
a=sendrecv
[Oct 30 12:57:16] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (553 bytes) to UDP:217.0.23.100:5060 --->
PRACK sip:sgc_c@217.0.23.100;transport=udp SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj2a1a23e9-d4f2-40ee-9dbb-35b93aa62981
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s1_3404367918-1
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20192 PRACK
Route: <sip:217.0.23.100;transport=udp;lr>
RAck: 2 20191 INVITE
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:16] VERBOSE[2460][C-00000001] app_dial.c: PJSIP/Telekom-00000001 is making progress passing it to PJSIP/300-00000000
[Oct 30 12:57:16] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (893 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---ce17818a508f924d
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
From: <sip:300@[IP-Adresse-Server]>;tag=3f03b043
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=07c0486a-39c9-4d99-a414-d512021fd037
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:[IP-Adresse-Server]:5060>
P-Asserted-Identity: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>
Content-Type: application/sdp
Content-Length: 270
v=0
o=- 0 2 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse-Server]
t=0 0
m=audio 30076 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:16] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (506 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj2a1a23e9-d4f2-40ee-9dbb-35b93aa62981
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s1_3404367918-1
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20192 PRACK
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, UPDATE, PRACK, INVITE, ACK, OPTIONS, CANCEL, BYE
[Oct 30 12:57:17] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (680 bytes) from UDP:[IP-Adresse-Client]:57980 --->
CANCEL sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---ce17818a508f924d
Max-Forwards: 70
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=3f03b043
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
CSeq: 2 CANCEL
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828631/2de0b28d60bfd381c5eb2d498690e707",uri="sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP",response="f4a801a34d781a83c92b3a67e1ccc475",cnonce="01de8a6f87a61caffe0c381b4055db75",nc=00000002,qop=auth,algorithm=md5,opaque="3826623d0d468ff3"
Content-Length: 0
[Oct 30 12:57:17] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (373 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---ce17818a508f924d
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
From: <sip:300@[IP-Adresse-Server]>;tag=3f03b043
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=07c0486a-39c9-4d99-a414-d512021fd037
CSeq: 2 CANCEL
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:17] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (555 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---ce17818a508f924d
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
From: <sip:300@[IP-Adresse-Server]>;tag=3f03b043
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=07c0486a-39c9-4d99-a414-d512021fd037
CSeq: 2 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>
Content-Length: 0
[Oct 30 12:57:17] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (444 bytes) to UDP:217.0.23.100:5060 --->
CANCEL sip:0[Telefonnummer Intern]@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj92b8f8ba-0f31-4126-93fa-8264aed9aa12
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20191 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:17] VERBOSE[2460][C-00000001] pbx.c: Spawn extension (internalsip, 0[Telefonnummer Intern], 1) exited non-zero on 'PJSIP/300-00000000'
[Oct 30 12:57:17] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (364 bytes) from UDP:[IP-Adresse-Client]:57980 --->
ACK sip:0[Telefonnummer Intern]@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---ce17818a508f924d
Max-Forwards: 70
To: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=07c0486a-39c9-4d99-a414-d512021fd037
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=3f03b043
Call-ID: 6DrtkFo9bQCSTnrTVAvYYA..
CSeq: 2 ACK
Content-Length: 0
[Oct 30 12:57:17] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (414 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj92b8f8ba-0f31-4126-93fa-8264aed9aa12
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s1_3404367918-1
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20191 CANCEL
Content-Length: 0
[Oct 30 12:57:17] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (530 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj92b8f8ba-0f31-4126-93fa-8264aed9aa12
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s1_3400723970-1344858605
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20191 INVITE
Contact: <sip:sgc_c@217.0.23.100;transport=udp>
Reason: Q.850;cause=17
Supported: timer
Content-Length: 0
[Oct 30 12:57:17] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (487 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:0[Telefonnummer Intern]@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj92b8f8ba-0f31-4126-93fa-8264aed9aa12
From: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=8d03f68e-81dd-455c-9c41-e3c812b83d73
To: <sip:0[Telefonnummer Intern]@tel.t-online.de>;tag=h7g4Esbg_p65563t1477828632m832251c566091993s1_3400723970-1344858605
Call-ID: 5c376a7d-398c-490f-be1c-c38a1cc14925
CSeq: 20191 ACK
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:19] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (949 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---89c0748a0412df42
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=cd211d71
Call-ID: 7cZqdDQdgPxIcvnGDe37Ig..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 260
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>
[Oct 30 12:57:19] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (522 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---89c0748a0412df42
Call-ID: 7cZqdDQdgPxIcvnGDe37Ig..
From: <sip:300@[IP-Adresse-Server]>;tag=cd211d71
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---89c0748a0412df42
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1477828639/bebd35369a1f0132a7326ef1bb0f2e7c",opaque="0774c4b95c3db623",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:19] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (698 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---38b889a51dba1da4
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=c004c420
Call-ID: RTeo6njj6e2fVvREpdBrJA..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:19] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (524 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---38b889a51dba1da4
Call-ID: RTeo6njj6e2fVvREpdBrJA..
From: <sip:300@[IP-Adresse-Server]>;tag=c004c420
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---38b889a51dba1da4
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1477828639/bebd35369a1f0132a7326ef1bb0f2e7c",opaque="1828d1f255ca49dc",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:19] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1246 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---9e7ffe4602c48c1f
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=cd211d71
Call-ID: 7cZqdDQdgPxIcvnGDe37Ig..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828639/bebd35369a1f0132a7326ef1bb0f2e7c",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="e5847e99561e375ad76a195f92cf64b1",cnonce="7cd9f3440f0e9acf10a6d11c055ed9d0",nc=00000001,qop=auth,algorithm=md5,opaque="0774c4b95c3db623"
Event: presence
Allow-Events: presence, kpml
Content-Length: 260
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>
[Oct 30 12:57:19] WARNING[2439] res_pjsip_pubsub.c: No registered publish handler for event presence
[Oct 30 12:57:19] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (372 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---9e7ffe4602c48c1f
Call-ID: 7cZqdDQdgPxIcvnGDe37Ig..
From: <sip:300@[IP-Adresse-Server]>;tag=cd211d71
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---9e7ffe4602c48c1f
CSeq: 2 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:19] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (995 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---b9eb1424269fff9c
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=c004c420
Call-ID: RTeo6njj6e2fVvREpdBrJA..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828639/bebd35369a1f0132a7326ef1bb0f2e7c",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="899ae01cbf255367181e8ba3097c9acd",cnonce="32d78e870a6016c11bb464889710b341",nc=00000001,qop=auth,algorithm=md5,opaque="1828d1f255ca49dc"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:19] WARNING[2439] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo
[Oct 30 12:57:19] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (374 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---b9eb1424269fff9c
Call-ID: RTeo6njj6e2fVvREpdBrJA..
From: <sip:300@[IP-Adresse-Server]>;tag=c004c420
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---b9eb1424269fff9c
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:23] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (494 bytes) to UDP:[IP-Adresse-Client]:57980 --->
OPTIONS sip:300@[IP-Adresse-Client]:57980;rinstance=ac8a4a8408b08df5 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPjffa9f07b-8ace-4f2d-8c48-58bd3e26210b
From: <sip:300@[IP-Adresse-Server]>;tag=d6609a21-3102-48ab-9f94-4ec66b7315ef
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>
Contact: <sip:300@[IP-Adresse-Server]:5060>
Call-ID: b53fa5fc-c115-487d-a605-8ad3929ddcd7
CSeq: 39633 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:23] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (685 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPjffa9f07b-8ace-4f2d-8c48-58bd3e26210b
Contact: <sip:[IP-Adresse-Client]:57980>
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=b069bc3f
From: <sip:300@[IP-Adresse-Server]>;tag=d6609a21-3102-48ab-9f94-4ec66b7315ef
Call-ID: b53fa5fc-c115-487d-a605-8ad3929ddcd7
CSeq: 39633 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:23] VERBOSE[2446] res_pjsip/pjsip_configuration.c: Contact 300/sip:300@[IP-Adresse-Client]:57980;rinstance=ac8a4a8408b08df5 is now Reachable. RTT: 10.635 msec
[Oct 30 12:57:23] VERBOSE[2446] res_pjsip/pjsip_configuration.c: Endpoint 300 is now Reachable
[Oct 30 12:57:25] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1247 bytes) from UDP:217.0.23.100:5060 --->
INVITE sip:0[Telefonnummer Intern]@[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 52
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iwoscovwg8j6fthp7v5e4mbrcn
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 1 INVITE
Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Min-Se: 900
P-Asserted-Identity: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>
Session-Expires: 1800
Supported: timer
Supported: histinfo
Supported: norefersub
Content-Type: application/sdp
Content-Length: 223
Session-ID: eb0ebffcf7003dd158ca9ad3eb001194
Request-Disposition: no-fork
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
v=0
o=- 1603706415 3413778947 IN IP4 217.0.23.100
s=-
c=IN IP4 217.0.4.103
t=0 0
m=audio 41044 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=maxptime:20
a=ptime:20
[Oct 30 12:57:25] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (491 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7iwoscovwg8j6fthp7v5e4mbrcn
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65563t1477828645m880346c566055813s2
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>
CSeq: 1 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:25] VERBOSE[2463][C-00000003] pbx.c: Executing [0[Telefonnummer Intern]@Telekom:1] Dial("PJSIP/Telekom-00000004", "PJSIP/300,15") in new stack
[Oct 30 12:57:25] VERBOSE[2463][C-00000003] app_dial.c: Called PJSIP/300
[Oct 30 12:57:25] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (1097 bytes) to UDP:[IP-Adresse-Client]:57980 --->
INVITE sip:300@[IP-Adresse-Client]:57980;rinstance=ac8a4a8408b08df5 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPj373bee90-a65f-4137-a6c1-d64f0a28c0fc
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>
Contact: <sip:asterisk@[IP-Adresse-Server]:5060>
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1679 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Type: application/sdp
Content-Length: 310
v=0
o=- 968728445 968728445 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse-Server]
t=0 0
m=audio 30002 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:25] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (346 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPj373bee90-a65f-4137-a6c1-d64f0a28c0fc
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1679 INVITE
Content-Length: 0
[Oct 30 12:57:25] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (432 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPj373bee90-a65f-4137-a6c1-d64f0a28c0fc
Contact: <sip:300@[IP-Adresse-Client]:57980>
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1679 INVITE
User-Agent: Z 3.9.32144 r32121
Content-Length: 0
[Oct 30 12:57:25] VERBOSE[2463][C-00000003] app_dial.c: PJSIP/300-00000005 is ringing
[Oct 30 12:57:25] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (680 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7iwoscovwg8j6fthp7v5e4mbrcn
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65563t1477828645m880346c566055813s2
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
CSeq: 1 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Contact: <sip:[IP-Adresse Nat Extern]:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (971 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPj373bee90-a65f-4137-a6c1-d64f0a28c0fc
Require: timer
Contact: <sip:300@[IP-Adresse-Client]:57980>
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1679 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 241
v=0
o=Z 0 2 IN IP4 [IP-Adresse-Client]
s=Z
c=IN IP4 [IP-Adresse-Client]
t=0 0
m=audio 8000 RTP/AVP 8 3 0 110 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (439 bytes) to UDP:[IP-Adresse-Client]:57980 --->
ACK sip:300@[IP-Adresse-Client]:57980 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPjb09e6461-ab9b-4c73-84da-5dd784b88760
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1679 ACK
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2463][C-00000003] app_dial.c: PJSIP/300-00000005 answered PJSIP/Telekom-00000004
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (1075 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7iwoscovwg8j6fthp7v5e4mbrcn
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65563t1477828645m880346c566055813s2
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
CSeq: 1 INVITE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:[IP-Adresse Nat Extern]:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 265
v=0
o=- 1603706415 3413778949 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse Nat Extern]
t=0 0
m=audio 30012 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (955 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---5e21882589a9df00
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=08651752
Call-ID: mFcDXF2Ych5oPKDNfY3nsQ..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (522 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---5e21882589a9df00
Call-ID: mFcDXF2Ych5oPKDNfY3nsQ..
From: <sip:300@[IP-Adresse-Server]>;tag=08651752
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---5e21882589a9df00
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1477828647/7f1f1d4b22f7f9e1ca1fb3c97218ea3c",opaque="0242d2b00a98760c",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (698 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---7fed87e5c595b5a9
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=9a07a844
Call-ID: 1knqikCuU6xRObgZI15avg..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (524 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---7fed87e5c595b5a9
Call-ID: 1knqikCuU6xRObgZI15avg..
From: <sip:300@[IP-Adresse-Server]>;tag=9a07a844
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---7fed87e5c595b5a9
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1477828647/7f1f1d4b22f7f9e1ca1fb3c97218ea3c",opaque="67567dd310c92ab8",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2464][C-00000003] bridge_channel.c: Channel PJSIP/300-00000005 joined 'simple_bridge' basic-bridge <e8903dc3-ced6-4443-8836-9dd096e13339>
[Oct 30 12:57:27] VERBOSE[2463][C-00000003] bridge_channel.c: Channel PJSIP/Telekom-00000004 joined 'simple_bridge' basic-bridge <e8903dc3-ced6-4443-8836-9dd096e13339>
[Oct 30 12:57:27] VERBOSE[2463][C-00000003] bridge.c: Bridge e8903dc3-ced6-4443-8836-9dd096e13339: switching from simple_bridge technology to native_rtp
[Oct 30 12:57:27] VERBOSE[2463][C-00000003] bridge_native_rtp.c: Remotely bridged 'PJSIP/Telekom-00000004' and 'PJSIP/300-00000005' - media will flow directly between them
[Oct 30 12:57:27] VERBOSE[2463][C-00000003] bridge_native_rtp.c: Remotely bridged 'PJSIP/Telekom-00000004' and 'PJSIP/300-00000005' - media will flow directly between them
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (1094 bytes) to UDP:[IP-Adresse-Client]:57980 --->
INVITE sip:300@[IP-Adresse-Client]:57980 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPj082151c0-919f-4e1b-ae09-112fda381024
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
Contact: <sip:asterisk@[IP-Adresse-Server]:5060>
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1680 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
P-Asserted-Identity: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Type: application/sdp
Content-Length: 307
v=0
o=- 968728445 968728446 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 217.0.4.103
t=0 0
m=audio 41044 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1252 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---4230723cbb3d851c
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=08651752
Call-ID: mFcDXF2Ych5oPKDNfY3nsQ..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828647/7f1f1d4b22f7f9e1ca1fb3c97218ea3c",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="4eb5003b94bd78ec608c6ddd0a63d6dd",cnonce="c0668ed278c2fda97f63cc715b49f875",nc=00000001,qop=auth,algorithm=md5,opaque="0242d2b00a98760c"
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[Oct 30 12:57:27] WARNING[2439] res_pjsip_pubsub.c: No registered publish handler for event presence
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (372 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---4230723cbb3d851c
Call-ID: mFcDXF2Ych5oPKDNfY3nsQ..
From: <sip:300@[IP-Adresse-Server]>;tag=08651752
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---4230723cbb3d851c
CSeq: 2 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (995 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---40cf65d8f9663e66
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=9a07a844
Call-ID: 1knqikCuU6xRObgZI15avg..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828647/7f1f1d4b22f7f9e1ca1fb3c97218ea3c",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="0a2558fc2e305ce4a6d5e8884af5be18",cnonce="d0f80acc52df6bcd5d9e38e45c9299a9",nc=00000001,qop=auth,algorithm=md5,opaque="67567dd310c92ab8"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:27] WARNING[2439] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (374 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---40cf65d8f9663e66
Call-ID: 1knqikCuU6xRObgZI15avg..
From: <sip:300@[IP-Adresse-Server]>;tag=9a07a844
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---40cf65d8f9663e66
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (971 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPj082151c0-919f-4e1b-ae09-112fda381024
Require: timer
Contact: <sip:300@[IP-Adresse-Client]:57980>
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1680 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 241
v=0
o=Z 0 3 IN IP4 [IP-Adresse-Client]
s=Z
c=IN IP4 [IP-Adresse-Client]
t=0 0
m=audio 8000 RTP/AVP 8 3 0 110 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (439 bytes) to UDP:[IP-Adresse-Client]:57980 --->
ACK sip:300@[IP-Adresse-Client]:57980 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPj468f300f-8f38-4d02-b8dc-d5427d8473c2
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1680 ACK
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (955 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---d3e842f1ffa9c094
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=3b4a9d11
Call-ID: H9hSiptD2rCv7z6YyLbVRg..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (522 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---d3e842f1ffa9c094
Call-ID: H9hSiptD2rCv7z6YyLbVRg..
From: <sip:300@[IP-Adresse-Server]>;tag=3b4a9d11
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---d3e842f1ffa9c094
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1477828647/7f1f1d4b22f7f9e1ca1fb3c97218ea3c",opaque="330d7c1455d47559",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (698 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---f7d38562ec642a69
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=752ab15b
Call-ID: kkW4qpErcmL9rMT32nJvNg..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (524 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---f7d38562ec642a69
Call-ID: kkW4qpErcmL9rMT32nJvNg..
From: <sip:300@[IP-Adresse-Server]>;tag=752ab15b
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---f7d38562ec642a69
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1477828647/7f1f1d4b22f7f9e1ca1fb3c97218ea3c",opaque="11b69dbf2e0963ce",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1252 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---cd5041188cedf172
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=3b4a9d11
Call-ID: H9hSiptD2rCv7z6YyLbVRg..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828647/7f1f1d4b22f7f9e1ca1fb3c97218ea3c",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="2d18d65d1c65225a541da8157877e9e5",cnonce="f39453d229229cfc074a80ffc26a60ed",nc=00000001,qop=auth,algorithm=md5,opaque="330d7c1455d47559"
Event: presence
Allow-Events: presence, kpml
Content-Length: 266
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>On the phone</note>
</tuple>
</presence>
[Oct 30 12:57:27] WARNING[2439] res_pjsip_pubsub.c: No registered publish handler for event presence
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (372 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---cd5041188cedf172
Call-ID: H9hSiptD2rCv7z6YyLbVRg..
From: <sip:300@[IP-Adresse-Server]>;tag=3b4a9d11
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---cd5041188cedf172
CSeq: 2 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (995 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---29f829208f713fbb
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=752ab15b
Call-ID: kkW4qpErcmL9rMT32nJvNg..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828647/7f1f1d4b22f7f9e1ca1fb3c97218ea3c",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="dd9ea8e3d056d5c9de20b966852d36af",cnonce="9b1573f69a5d9ac672d1551f6d2ada98",nc=00000001,qop=auth,algorithm=md5,opaque="11b69dbf2e0963ce"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:27] WARNING[2439] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (374 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---29f829208f713fbb
Call-ID: kkW4qpErcmL9rMT32nJvNg..
From: <sip:300@[IP-Adresse-Server]>;tag=752ab15b
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---29f829208f713fbb
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (545 bytes) from UDP:217.0.23.100:5060 --->
ACK sip:[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 54
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7irj4feerj4qccc115g4xwybvg9
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 1 ACK
Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (1151 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:sgc_c@217.0.23.100;transport=udp SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj1e1fb87c-fb65-4ef3-bd04-016aeca203a8
From: <sip:0[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
To: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Contact: <sip:[IP-Adresse Nat Extern]:5060>
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11161 INVITE
Route: <sip:217.0.23.100;transport=udp;lr>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Type: application/sdp
Content-Length: 311
v=0
o=- 1603706415 3413778950 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse Nat Extern]
t=0 0
m=audio 8000 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (609 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 407 Proxy Authentication Required 02035034C
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPj1e1fb87c-fb65-4ef3-bd04-016aeca203a8
To: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
From: <sip:0[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11161 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="DFA5356332E0155800000000ABAA072F",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (556 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:sgc_c@217.0.23.100;transport=udp SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj1e1fb87c-fb65-4ef3-bd04-016aeca203a8
From: <sip:0[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
To: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11161 ACK
Route: <sip:217.0.23.100;transport=udp;lr>
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (1444 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:sgc_c@217.0.23.100;transport=udp SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPjd8db4702-2a3a-4f8f-a935-9cee985b249a
From: <sip:0[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
To: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Contact: <sip:[IP-Adresse Nat Extern]:5060>
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11162 INVITE
Route: <sip:217.0.23.100;transport=udp;lr>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Proxy-Authorization: Digest username="0[Telefonnummer Intern]", realm="tel.t-online.de", nonce="DFA5356332E0155800000000ABAA072F", uri="sip:sgc_c@217.0.23.100;transport=udp", response="4adad67a2ee16f4ec671a0a95feffc49", algorithm=MD5, cnonce="4a475811-75ea-4c25-a0ab-6f2ffcb187ab", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 311
v=0
o=- 1603706415 3413778950 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse Nat Extern]
t=0 0
m=audio 8000 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:27] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (1164 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;received=[IP-Adresse Nat Extern];rport=5060;branch=z9hG4bKPjd8db4702-2a3a-4f8f-a935-9cee985b249a
To: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
From: <sip:0[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11162 INVITE
Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Session-Expires: 1800;refresher=uas
Supported: timer
Supported: 100rel
Supported: histinfo
Supported: norefersub
Content-Type: application/sdp
Content-Length: 229
Authentication-Info: qop=auth,rspauth="778348c67c9508e96dc0cd4bf66fed95",cnonce="4a475811-75ea-4c25-a0ab-6f2ffcb187ab",nc=00000001
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
v=0
o=- 1603706415 3413778948 IN IP4 217.0.23.100
s=-
c=IN IP4 217.0.4.103
t=0 0
m=audio 41044 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=ptime:20
a=sendrecv
[Oct 30 12:57:27] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (556 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:sgc_c@217.0.23.100;transport=udp SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse Nat Extern]:5060;rport;branch=z9hG4bKPj36f3e2fd-8ec9-483a-b8f5-37c6fe6cc330
From: <sip:0[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
To: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11162 ACK
Route: <sip:217.0.23.100;transport=udp;lr>
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:38] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (447 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ir7eh12grx58ryxy23w689jdt6
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11163 OPTIONS
Content-Length: 0
[Oct 30 12:57:38] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (538 bytes) from UDP:217.0.23.100:5060 --->
BYE sip:[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 54
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ifybmnyh69xdjlxb8qv2gsvfwz
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11164 BYE
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
[Oct 30 12:57:38] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (478 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7ifybmnyh69xdjlxb8qv2gsvfwz
Call-ID: p65563t1477828645m880346c566055813s2
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
CSeq: 11164 BYE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:38] VERBOSE[2463][C-00000003] bridge_channel.c: Channel PJSIP/Telekom-00000004 left 'native_rtp' basic-bridge <e8903dc3-ced6-4443-8836-9dd096e13339>
[Oct 30 12:57:38] VERBOSE[2463][C-00000003] pbx.c: Spawn extension (Telekom, 0[Telefonnummer Intern], 1) exited non-zero on 'PJSIP/Telekom-00000004'
[Oct 30 12:57:38] VERBOSE[2464][C-00000003] bridge_channel.c: Channel PJSIP/300-00000005 left 'native_rtp' basic-bridge <e8903dc3-ced6-4443-8836-9dd096e13339>
[Oct 30 12:57:38] VERBOSE[2440] res_pjsip_logger.c: <--- Transmitting SIP request (1097 bytes) to UDP:[IP-Adresse-Client]:57980 --->
INVITE sip:300@[IP-Adresse-Client]:57980 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPj56e397f9-407e-4f47-bfa7-dd1e1e07ed93
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
Contact: <sip:asterisk@[IP-Adresse-Server]:5060>
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1681 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
P-Asserted-Identity: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Type: application/sdp
Content-Length: 310
v=0
o=- 968728445 968728447 IN IP4 [IP-Adresse-Server]
s=Asterisk
c=IN IP4 [IP-Adresse-Server]
t=0 0
m=audio 30002 RTP/AVP 9 8 0 3 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[Oct 30 12:57:38] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (447 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ir7eh12grx58ryxy23w689jdt6
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11163 OPTIONS
Content-Length: 0
[Oct 30 12:57:38] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (971 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPj56e397f9-407e-4f47-bfa7-dd1e1e07ed93
Require: timer
Contact: <sip:300@[IP-Adresse-Client]:57980>
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1681 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 241
v=0
o=Z 0 4 IN IP4 [IP-Adresse-Client]
s=Z
c=IN IP4 [IP-Adresse-Client]
t=0 0
m=audio 8000 RTP/AVP 8 3 0 110 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[Oct 30 12:57:38] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (439 bytes) to UDP:[IP-Adresse-Client]:57980 --->
ACK sip:300@[IP-Adresse-Client]:57980 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPjfab06fe4-45c0-47f2-a838-c5ef45529d77
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1681 ACK
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:38] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP request (439 bytes) to UDP:[IP-Adresse-Client]:57980 --->
BYE sip:300@[IP-Adresse-Client]:57980 SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport;branch=z9hG4bKPj31cca512-e8ed-4b01-80b0-56c50fd28a09
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1682 BYE
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:38] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP response (424 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-Adresse-Server]:5060;rport=5060;branch=z9hG4bKPj31cca512-e8ed-4b01-80b0-56c50fd28a09
Contact: <sip:300@[IP-Adresse-Client]:57980>
To: <sip:300@[IP-Adresse-Client];rinstance=ac8a4a8408b08df5>;tag=333c1379
From: <sip:0[Telefonnummer Intern]@[IP-Adresse-Server]>;tag=0fd39ae8-b559-495e-b1a0-6de89e2f01d9
Call-ID: 283ca346-02d2-45fa-8393-4c5d02fc079e
CSeq: 1682 BYE
User-Agent: Z 3.9.32144 r32121
Content-Length: 0
[Oct 30 12:57:39] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (447 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ir7eh12grx58ryxy23w689jdt6
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11163 OPTIONS
Content-Length: 0
[Oct 30 12:57:41] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (949 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---ec1e965707c831c4
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=8707115f
Call-ID: z4Qe8P9aA8cA2s8LJD5pAg..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 260
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>
[Oct 30 12:57:41] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (522 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---ec1e965707c831c4
Call-ID: z4Qe8P9aA8cA2s8LJD5pAg..
From: <sip:300@[IP-Adresse-Server]>;tag=8707115f
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---ec1e965707c831c4
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1477828661/d2b9315606e6fd537181a6a2b1ee7ac9",opaque="12b53b47295ecb9d",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:41] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (698 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---d1f4bfbdbbddef3c
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=5656e568
Call-ID: JkiIZPyjEcSAeiAXSvbNkg..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:41] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (524 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---d1f4bfbdbbddef3c
Call-ID: JkiIZPyjEcSAeiAXSvbNkg..
From: <sip:300@[IP-Adresse-Server]>;tag=5656e568
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---d1f4bfbdbbddef3c
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1477828661/d2b9315606e6fd537181a6a2b1ee7ac9",opaque="55ae588d029ddcf5",algorithm=md5,qop="auth"
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:41] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (1246 bytes) from UDP:[IP-Adresse-Client]:57980 --->
PUBLISH sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---3212bc54fb006196
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=8707115f
Call-ID: z4Qe8P9aA8cA2s8LJD5pAg..
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828661/d2b9315606e6fd537181a6a2b1ee7ac9",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="eab54bfcad0525155b3abe6a7a892dc2",cnonce="2877799689f8b4b0e4a430120ff65c54",nc=00000001,qop=auth,algorithm=md5,opaque="12b53b47295ecb9d"
Event: presence
Allow-Events: presence, kpml
Content-Length: 260
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:300@[IP-Adresse-Server];transport=UDP">
<tuple id="300" >
<status><basic>open</basic></status>
<note>Online</note>
</tuple>
</presence>
[Oct 30 12:57:41] WARNING[2439] res_pjsip_pubsub.c: No registered publish handler for event presence
[Oct 30 12:57:41] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (372 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---3212bc54fb006196
Call-ID: z4Qe8P9aA8cA2s8LJD5pAg..
From: <sip:300@[IP-Adresse-Server]>;tag=8707115f
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---3212bc54fb006196
CSeq: 2 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:41] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (995 bytes) from UDP:[IP-Adresse-Client]:57980 --->
SUBSCRIBE sip:300@[IP-Adresse-Server];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;branch=z9hG4bK-524287-1---3e743de82731904f
Max-Forwards: 70
Contact: <sip:300@[IP-Adresse-Client]:57980;transport=UDP>
To: <sip:300@[IP-Adresse-Server];transport=UDP>
From: <sip:300@[IP-Adresse-Server];transport=UDP>;tag=5656e568
Call-ID: JkiIZPyjEcSAeiAXSvbNkg..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="300",realm="asterisk",nonce="1477828661/d2b9315606e6fd537181a6a2b1ee7ac9",uri="sip:300@[IP-Adresse-Server];transport=UDP",response="88d1efcd39b3ec54e9890ca0889f0bbd",cnonce="fc86dc2c859ae04d80535cf16362b0de",nc=00000001,qop=auth,algorithm=md5,opaque="55ae588d029ddcf5"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
[Oct 30 12:57:41] WARNING[2439] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo
[Oct 30 12:57:41] VERBOSE[2439] res_pjsip_logger.c: <--- Transmitting SIP response (374 bytes) to UDP:[IP-Adresse-Client]:57980 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [IP-Adresse-Client]:57980;rport=57980;received=[IP-Adresse-Client];branch=z9hG4bK-524287-1---3e743de82731904f
Call-ID: JkiIZPyjEcSAeiAXSvbNkg..
From: <sip:300@[IP-Adresse-Server]>;tag=5656e568
To: <sip:300@[IP-Adresse-Server]>;tag=z9hG4bK-524287-1---3e743de82731904f
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.4(13.9.1)
Content-Length: 0
[Oct 30 12:57:41] VERBOSE[2438] res_pjsip_logger.c: <--- Received SIP request (447 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:[IP-Adresse Nat Extern]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ir7eh12grx58ryxy23w689jdt6
To: <sip:+49[Telefonnummer Intern]@tel.t-online.de;user=phone>;tag=8bf3098a-fe56-4b7f-bacd-00f09cd01132
From: <sip:+49[Telefonnummer Extern]@tmobile.de;user=phone>;tag=h7g4Esbg_p65563t1477828645m880346c566055813s1_3413779196-1645404730
Call-ID: p65563t1477828645m880346c566055813s2
CSeq: 11163 OPTIONS
Content-Length: 0
Du forwardest ja wieder eine interne Nummer per SIP an den Provider: sip:300@[IP-Adresse-Server
Wie der Lord oben schon mehrfach gesagt hat kann das ja niemals funktionieren, denn die Telekomm wird nur einen SIP Request auf dir zugeteilte Nummern zulassen, niemals auf interne logischerweise.
Mit einem Wireshark Trace kannst du doch sofort sehen was da per SIP Richtung Provider geht und auch was genau der antwortet.
Das sagt doch eigentlich schon alles....
Wie der Lord oben schon mehrfach gesagt hat kann das ja niemals funktionieren, denn die Telekomm wird nur einen SIP Request auf dir zugeteilte Nummern zulassen, niemals auf interne logischerweise.
Mit einem Wireshark Trace kannst du doch sofort sehen was da per SIP Richtung Provider geht und auch was genau der antwortet.
Das sagt doch eigentlich schon alles....
Das rein und raustelefonieren funktioniert jetzt, bekomme aber keinen Ton.
Woran kann das liegen, bzw was braucht Ihr damit Ihr mir helfen könnt?
Woran kann das liegen, bzw was braucht Ihr damit Ihr mir helfen könnt?
Meistens liegt es an einem Problem mit der Portweiterleitung...
Das ist eine häufige Ursache für fehlenden Ton bei SIP-Telefonie...
Das ist eine häufige Ursache für fehlenden Ton bei SIP-Telefonie...
daran kann es nicht liegen. Ich hatte vorher eine funktionieren Config mit Asterisk und FreePBX, da ich aber bestimmte Anderungen nicht in FreePBX vornhemen kann, möchte die "pur" Asterisk nutzen.
Es muss ein Config-Fehler sein, ich weiß aber nicht wo. pjsip- und rtp-Config habe ich kopiert
Es muss ein Config-Fehler sein, ich weiß aber nicht wo. pjsip- und rtp-Config habe ich kopiert
Vermutlich versucht die Asterisk ein Direct Bridging mit dem Client hinter dem NAT.
Ich kenne jetzt pjsip nicht ganz auswendig (der Library bringe ich momentan noch großes Misstrauen entgegen).
Im klassischen SIP-Channeltreiber sollte als globale Konfiguration hinterlegt sein, dass die Asterisk immer NAT annimmt (nat=force_rport,comedia) und niemals versucht die Audiokanäle direkt zwischen den Teilnehmern zu bridgen:
Wie das in pjsip konfiguriert wird, kann ich dir jetzt leider aus dem Kopf nicht beantworten, vermutlich sind folgende Konfigurationen notwendig:
Mit "rtp_symmetric = yes" solltest du dir sämtliche Port-Forwardings sparen können, da hierbei für ein- und ausgehenden Traffic der gleiche Port verwendet wird und ein NAT-Router somit die eingehenden Pakete als "established connection" erkennt und durchlässt.
Natürlich unter der Prämisse, dass da keine Stateless-Firewall läuft, der explizit Portranges erlaubt werden müssen - das müsste dann weiterhin getan werden. An einem "Standard"-Heimnetz-Router ist das allerdings nicht notwendig.
Die Konfiguration im zweiten Code-Block gehört wie gesagt in einen globalen Teil der SIP-Config, weil diese Konfiguration allgemein so funktionieren dürfte. Und selbst ohne NAT tut es nicht weh, wenn der Server davon ausgeht dass er nicht einfach alle Ports benutzen kann
Ggf. ist es noch notwendig, der Telekom-Konfiguration ein " ice_support = yes " zu spendieren, um NAT-Traversal zu aktivieren und die öffentliche IP-Adresse in den SDP-Teil der INVITES zu schreiben.
Ich kenne jetzt pjsip nicht ganz auswendig (der Library bringe ich momentan noch großes Misstrauen entgegen).
Im klassischen SIP-Channeltreiber sollte als globale Konfiguration hinterlegt sein, dass die Asterisk immer NAT annimmt (nat=force_rport,comedia) und niemals versucht die Audiokanäle direkt zwischen den Teilnehmern zu bridgen:
nat=force_port,comedia
directmedia=no
directrtpsetup=no
canreinvite=noWie das in pjsip konfiguriert wird, kann ich dir jetzt leider aus dem Kopf nicht beantworten, vermutlich sind folgende Konfigurationen notwendig:
direct_media = no
disable_direct_media_on_nat = yes (wer da den Default "no" eingeführt hat, gehört mit einem NAT-Router verdroschen!)
rtp_symmetric = yesMit "rtp_symmetric = yes" solltest du dir sämtliche Port-Forwardings sparen können, da hierbei für ein- und ausgehenden Traffic der gleiche Port verwendet wird und ein NAT-Router somit die eingehenden Pakete als "established connection" erkennt und durchlässt.
Natürlich unter der Prämisse, dass da keine Stateless-Firewall läuft, der explizit Portranges erlaubt werden müssen - das müsste dann weiterhin getan werden. An einem "Standard"-Heimnetz-Router ist das allerdings nicht notwendig.
Die Konfiguration im zweiten Code-Block gehört wie gesagt in einen globalen Teil der SIP-Config, weil diese Konfiguration allgemein so funktionieren dürfte. Und selbst ohne NAT tut es nicht weh, wenn der Server davon ausgeht dass er nicht einfach alle Ports benutzen kann
Ggf. ist es noch notwendig, der Telekom-Konfiguration ein " ice_support = yes " zu spendieren, um NAT-Traversal zu aktivieren und die öffentliche IP-Adresse in den SDP-Teil der INVITES zu schreiben.
Ich danke dir für die Erleichtung, genau das hat gefehlt. 
Es funktioniert jetzt einwandfrei.
Nochmals vielen Danke für eure Hilfe !!!
Es funktioniert jetzt einwandfrei.
Nochmals vielen Danke für eure Hilfe !!!
