hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is?> Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for > details. > This is free software, with components licensed under the GNU General > Public > License version 2 and other licenses; you are welcome to redistribute it > under > certain conditions. Type 'core show license' for details. > ========================================================================> == Parsing '/etc/asterisk/asterisk.conf': Found > Connected to Asterisk 1.4.21.2 currently running on gd-branch (pid = 3145) > Verbosity is at least 3 > -- Executing [015921256331 at from-internal:1] Set("SIP/75002-b7705298", > "MOHCLASS=none") in new stack > -- Executing [015921256331 at from-internal:2] > Macro("SIP/75002-b7705298", "user-callerid|SKIPTTL|") in new stack > -- Executing [s at macro-user-callerid:1] Set("SIP/75002-b7705298", > "AMPUSER=75002") in new stack > -- Executing [s at macro-user-callerid:2] GotoIf("SIP/75002-b7705298", > "0?report") in new stack > -- Executing [s at macro-user-callerid:3] ExecIf("SIP/75002-b7705298", > "1|Set|REALCALLERIDNUM=75002") in new stack > -- Executing [s at macro-user-callerid:4] Set("SIP/75002-b7705298", > "AMPUSER=75002") in new stack > -- Executing [s at macro-user-callerid:5] Set("SIP/75002-b7705298", > "AMPUSERCIDNAME=75002") in new stack > -- Executing [s at macro-user-callerid:6] GotoIf("SIP/75002-b7705298", > "0?report") in new stack > -- Executing [s at macro-user-callerid:7] Set("SIP/75002-b7705298", > "AMPUSERCID=75002") in new stack > -- Executing [s at macro-user-callerid:8] Set("SIP/75002-b7705298", > "CALLERID(all)="75002" <75002>") in new stack > -- Executing [s at macro-user-callerid:9] ExecIf("SIP/75002-b7705298", > "0|Set|CHANNEL(language)=") in new stack > -- Executing [s at macro-user-callerid:10] GotoIf("SIP/75002-b7705298", > "1?continue") in new stack > -- Goto (macro-user-callerid,s,19) > -- Executing [s at macro-user-callerid:19] NoOp("SIP/75002-b7705298", > "Using CallerID "75002" <75002>") in new stack > -- Executing [015921256331 at from-internal:3] Set("SIP/75002-b7705298", > "_NODEST=") in new stack > -- Executing [015921256331 at from-internal:4] > Macro("SIP/75002-b7705298", "record-enable|75002|OUT|") in new stack > -- Executing [s at macro-record-enable:1] GotoIf("SIP/75002-b7705298", > "1?check") in new stack > -- Goto (macro-record-enable,s,4) > -- Executing [s at macro-record-enable:4] AGI("SIP/75002-b7705298", > "recordingcheck|20100326-141638|1269584198.62") in new stack > -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck > recordingcheck|20100326-141638|1269584198.62: Outbound recording enabled. > recordingcheck|20100326-141638|1269584198.62: > CALLFILENAME=OUT75002-20100326-141638-1269584198.62 > -- AGI Script recordingcheck completed, returning 0 > -- Executing [s at macro-record-enable:999] > MixMonitor("SIP/75002-b7705298", > "/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav||") > in new stack > -- Executing [s at macro-record-enable:1000] Set("SIP/75002-b7705298", > "RecordingFileName=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") > in new stack > -- Executing [s at macro-record-enable:1001] NoOp("SIP/75002-b7705298", > "/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") > in new stack > -- Executing [s at macro-record-enable:1002] Set("SIP/75002-b7705298", > "CDR(userfield)=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") > in new stack > -- Executing [015921256331 at from-internal:5] > Macro("SIP/75002-b7705298", "dialout-trunk|7|015921256331||") in new stack > -- Executing [s at macro-dialout-trunk:1] Set("SIP/75002-b7705298", > "DIAL_TRUNK=7") in new stack > -- Executing [s at macro-dialout-trunk:2] GosubIf("SIP/75002-b7705298", > "0?sub-pincheck|s|1") in new stack > -- Executing [s at macro-dialout-trunk:3] GotoIf("SIP/75002-b7705298", > "0?disabletrunk|1") in new stack > -- Executing [s at macro-dialout-trunk:4] Set("SIP/75002-b7705298", > "DIAL_NUMBER=015921256331") in new stack > -- Executing [s at macro-dialout-trunk:5] Set("SIP/75002-b7705298", > "DIAL_TRUNK_OPTIONS=Ttr") in new stack > -- Executing [s at macro-dialout-trunk:6] Set("SIP/75002-b7705298", > "OUTBOUND_GROUP=OUT_7") in new stack > -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/75002-b7705298", > "1?nomax") in new stack > -- Goto (macro-dialout-trunk,s,9) > -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/75002-b7705298", > "0?skipoutcid") in new stack > -- Executing [s at macro-dialout-trunk:10] Set("SIP/75002-b7705298", > "DIAL_TRUNK_OPTIONS=Tt") in new stack > == Begin MixMonitor Recording SIP/75002-b7705298 > -- Executing [s at macro-dialout-trunk:11] Macro("SIP/75002-b7705298", > "outbound-callerid|7") in new stack > -- Executing [s at macro-outbound-callerid:1] > ExecIf("SIP/75002-b7705298", "0|SetCallerPres|") in new stack > -- Executing [s at macro-outbound-callerid:2] > ExecIf("SIP/75002-b7705298", "0|Set|REALCALLERIDNUM=75002") in new stack > -- Executing [s at macro-outbound-callerid:3] > GotoIf("SIP/75002-b7705298", "1?normcid") in new stack > -- Goto (macro-outbound-callerid,s,6) > -- Executing [s at macro-outbound-callerid:6] Set("SIP/75002-b7705298", > "USEROUTCID=") in new stack > -- Executing [s at macro-outbound-callerid:7] Set("SIP/75002-b7705298", > "EMERGENCYCID=") in new stack > -- Executing [s at macro-outbound-callerid:8] Set("SIP/75002-b7705298", > "TRUNKOUTCID=s2") in new stack > -- Executing [s at macro-outbound-callerid:9] > GotoIf("SIP/75002-b7705298", "1?trunkcid") in new stack > -- Goto (macro-outbound-callerid,s,12) > -- Executing [s at macro-outbound-callerid:12] > ExecIf("SIP/75002-b7705298", "1|Set|CALLERID(all)=s2") in new stack > -- Executing [s at macro-outbound-callerid:13] > ExecIf("SIP/75002-b7705298", "0|Set|CALLERID(all)=") in new stack > -- Executing [s at macro-outbound-callerid:14] > ExecIf("SIP/75002-b7705298", "0|SetCallerPres|prohib_passed_screen") in new > stack > -- Executing [s at macro-dialout-trunk:12] ExecIf("SIP/75002-b7705298", > "0|AGI|fixlocalprefix") in new stack > -- Executing [s at macro-dialout-trunk:13] Set("SIP/75002-b7705298", > "OUTNUM=015921256331") in new stack > -- Executing [s at macro-dialout-trunk:14] Set("SIP/75002-b7705298", > "custom=SIP/s2") in new stack > -- Executing [s at macro-dialout-trunk:15] ExecIf("SIP/75002-b7705298", > "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack > -- Executing [s at macro-dialout-trunk:16] Macro("SIP/75002-b7705298", > "dialout-trunk-predial-hook|") in new stack > -- Executing [s at macro-dialout-trunk-predial-hook:1] > MacroExit("SIP/75002-b7705298", "") in new stack > -- Executing [s at macro-dialout-trunk:17] GotoIf("SIP/75002-b7705298", > "0?bypass|1") in new stack > -- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/75002-b7705298", > "0?customtrunk") in new stack > -- Executing [s at macro-dialout-trunk:19] Dial("SIP/75002-b7705298", > "SIP/s2/015921256331|300|M(setmusic^none)Tt") in new stack > Audio is at 219.235.234.238 port 17136 > Adding codec 0x4 (ulaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to 222.46.18.52:5060: > INVITE sip:015921256331 at 222.46.18.52 <sip%3A015921256331 at 222.46.18.52>SIP/2.0 > Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport > From: "s2" <sip:Unknown at 222.46.18.52 <sip%3AUnknown at 222.46.18.52> > >;tag=as75543a2d > To: <sip:015921256331 at 222.46.18.52 <sip%3A015921256331 at 222.46.18.52>> > Contact: <sip:Unknown at 219.235.234.238 <sip%3AUnknown at 219.235.234.238>> > Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Fri, 26 Mar 2010 06:16:38 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 244 > > v=0 > o=root 3145 3145 IN IP4 219.235.234.238 > s=session > c=IN IP4 219.235.234.238 > t=0 0 > m=audio 17136 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- > -- Called s2/015921256331 > gd-branch*CLI> > <--- SIP read from 222.46.18.52:5060 ---> > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP 219.235.234.238:5060 > ;branch=z9hG4bK368b5ad8;received=58.247.12.18;rport=11028 > From: "s2" <sip:Unknown at 222.46.18.52 <sip%3AUnknown at 222.46.18.52> > >;tag=as75543a2d > To: <sip:015921256331 at 222.46.18.52 <sip%3A015921256331 at 222.46.18.52>> > Contact: <sip:015921256331 at 222.46.18.52:5060> > Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52 > CSeq: 102 INVITE > Max-Forwards: 70 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: timer > Server: VOS2009 V2.1.1.8 > > > <-------------> > --- (11 headers 0 lines) --- > Transmitting (NAT) to 222.46.18.52:5060: > ACK sip:015921256331 at 222.46.18.52 <sip%3A015921256331 at 222.46.18.52>SIP/2.0 > Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport > From: "s2" <sip:Unknown at 222.46.18.52 <sip%3AUnknown at 222.46.18.52> > >;tag=as75543a2d > To: <sip:015921256331 at 222.46.18.52 <sip%3A015921256331 at 222.46.18.52>> > Contact: <sip:Unknown at 219.235.234.238 <sip%3AUnknown at 219.235.234.238>> > Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > > --- > -- SIP/s2-088f72e8 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing [s at macro-dialout-trunk:20] Goto("SIP/75002-b7705298", > "s-CONGESTION|1") in new stack > -- Goto (macro-dialout-trunk,s-CONGESTION,1) > -- Executing [s-CONGESTION at macro-dialout-trunk:1] > GotoIf("SIP/75002-b7705298", "1?noreport") in new stack > -- Goto (macro-dialout-trunk,s-CONGESTION,3) > -- Executing [s-CONGESTION at macro-dialout-trunk:3] > NoOp("SIP/75002-b7705298", "TRUNK Dial failed due to CONGESTION - failing > through to other trunks") in new stack > -- Executing [015921256331 at from-internal:6] > Macro("SIP/75002-b7705298", "outisbusy|") in new stack > -- Executing [s at macro-outisbusy:1] Playback("SIP/75002-b7705298", > "all-circuits-busy-now|noanswer") in new stack > -- <SIP/75002-b7705298> Playing 'all-circuits-busy-now' (language 'en') > Really destroying SIP dialog > '5cf71e106209cf65344e24031354fbda at 222.46.18.52' Method: INVITE > -- Executing [s at macro-outisbusy:2] Playback("SIP/75002-b7705298", > "pls-try-call-later|noanswer") in new stack > -- <SIP/75002-b7705298> Playing 'pls-try-call-later' (language 'en') > -- Executing [s at macro-outisbusy:3] Macro("SIP/75002-b7705298", > "hangupcall") in new stack > -- Executing [s at macro-hangupcall:1] GotoIf("SIP/75002-b7705298", > "1?skiprg") in new stack > -- Goto (macro-hangupcall,s,4) > -- Executing [s at macro-hangupcall:4] GotoIf("SIP/75002-b7705298", > "1?skipblkvm") in new stack > -- Goto (macro-hangupcall,s,7) > -- Executing [s at macro-hangupcall:7] GotoIf("SIP/75002-b7705298", > "1?theend") in new stack > -- Goto (macro-hangupcall,s,9) > -- Executing [s at macro-hangupcall:9] Hangup("SIP/75002-b7705298", "") > in new stack > == Spawn extension (macro-hangupcall, s, 9) exited non-zero on > 'SIP/75002-b7705298' in macro 'hangupcall' > == Spawn extension (macro-hangupcall, s, 9) exited non-zero on > 'SIP/75002-b7705298' in macro 'outisbusy' > == Spawn extension (macro-hangupcall, s, 9) exited non-zero on > 'SIP/75002-b7705298' > == End MixMonitor Recording SIP/75002-b7705298 >-- Best Regards! Aaron Chen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/84c6d1e3/attachment.htm
You need to ask your carrier what you are not sending them that they would like. It's usually a fromdomain or authname. ----- Original Message ----- From: Aaron chen To: Asterisk Users Mailing List - Non-Commercial Discussion ; Asterisk Developers Mailing List Sent: Friday, March 26, 2010 09:22 Subject: [asterisk-users] SIP/2.0 403 Forbidden hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ======================================================================== == Parsing '/etc/asterisk/asterisk.conf': Found Connected to Asterisk 1.4.21.2 currently running on gd-branch (pid = 3145) Verbosity is at least 3 -- Executing [015921256331 at from-internal:1] Set("SIP/75002-b7705298", "MOHCLASS=none") in new stack -- Executing [015921256331 at from-internal:2] Macro("SIP/75002-b7705298", "user-callerid|SKIPTTL|") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/75002-b7705298", "AMPUSER=75002") in new stack -- Executing [s at macro-user-callerid:2] GotoIf("SIP/75002-b7705298", "0?report") in new stack -- Executing [s at macro-user-callerid:3] ExecIf("SIP/75002-b7705298", "1|Set|REALCALLERIDNUM=75002") in new stack -- Executing [s at macro-user-callerid:4] Set("SIP/75002-b7705298", "AMPUSER=75002") in new stack -- Executing [s at macro-user-callerid:5] Set("SIP/75002-b7705298", "AMPUSERCIDNAME=75002") in new stack -- Executing [s at macro-user-callerid:6] GotoIf("SIP/75002-b7705298", "0?report") in new stack -- Executing [s at macro-user-callerid:7] Set("SIP/75002-b7705298", "AMPUSERCID=75002") in new stack -- Executing [s at macro-user-callerid:8] Set("SIP/75002-b7705298", "CALLERID(all)="75002" <75002>") in new stack -- Executing [s at macro-user-callerid:9] ExecIf("SIP/75002-b7705298", "0|Set|CHANNEL(language)=") in new stack -- Executing [s at macro-user-callerid:10] GotoIf("SIP/75002-b7705298", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s at macro-user-callerid:19] NoOp("SIP/75002-b7705298", "Using CallerID "75002" <75002>") in new stack -- Executing [015921256331 at from-internal:3] Set("SIP/75002-b7705298", "_NODEST=") in new stack -- Executing [015921256331 at from-internal:4] Macro("SIP/75002-b7705298", "record-enable|75002|OUT|") in new stack -- Executing [s at macro-record-enable:1] GotoIf("SIP/75002-b7705298", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s at macro-record-enable:4] AGI("SIP/75002-b7705298", "recordingcheck|20100326-141638|1269584198.62") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20100326-141638|1269584198.62: Outbound recording enabled. recordingcheck|20100326-141638|1269584198.62: CALLFILENAME=OUT75002-20100326-141638-1269584198.62 -- AGI Script recordingcheck completed, returning 0 -- Executing [s at macro-record-enable:999] MixMonitor("SIP/75002-b7705298", "/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav||") in new stack -- Executing [s at macro-record-enable:1000] Set("SIP/75002-b7705298", "RecordingFileName=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") in new stack -- Executing [s at macro-record-enable:1001] NoOp("SIP/75002-b7705298", "/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") in new stack -- Executing [s at macro-record-enable:1002] Set("SIP/75002-b7705298", "CDR(userfield)=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav") in new stack -- Executing [015921256331 at from-internal:5] Macro("SIP/75002-b7705298", "dialout-trunk|7|015921256331||") in new stack -- Executing [s at macro-dialout-trunk:1] Set("SIP/75002-b7705298", "DIAL_TRUNK=7") in new stack -- Executing [s at macro-dialout-trunk:2] GosubIf("SIP/75002-b7705298", "0?sub-pincheck|s|1") in new stack -- Executing [s at macro-dialout-trunk:3] GotoIf("SIP/75002-b7705298", "0?disabletrunk|1") in new stack -- Executing [s at macro-dialout-trunk:4] Set("SIP/75002-b7705298", "DIAL_NUMBER=015921256331") in new stack -- Executing [s at macro-dialout-trunk:5] Set("SIP/75002-b7705298", "DIAL_TRUNK_OPTIONS=Ttr") in new stack -- Executing [s at macro-dialout-trunk:6] Set("SIP/75002-b7705298", "OUTBOUND_GROUP=OUT_7") in new stack -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/75002-b7705298", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/75002-b7705298", "0?skipoutcid") in new stack -- Executing [s at macro-dialout-trunk:10] Set("SIP/75002-b7705298", "DIAL_TRUNK_OPTIONS=Tt") in new stack == Begin MixMonitor Recording SIP/75002-b7705298 -- Executing [s at macro-dialout-trunk:11] Macro("SIP/75002-b7705298", "outbound-callerid|7") in new stack -- Executing [s at macro-outbound-callerid:1] ExecIf("SIP/75002-b7705298", "0|SetCallerPres|") in new stack -- Executing [s at macro-outbound-callerid:2] ExecIf("SIP/75002-b7705298", "0|Set|REALCALLERIDNUM=75002") in new stack -- Executing [s at macro-outbound-callerid:3] GotoIf("SIP/75002-b7705298", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s at macro-outbound-callerid:6] Set("SIP/75002-b7705298", "USEROUTCID=") in new stack -- Executing [s at macro-outbound-callerid:7] Set("SIP/75002-b7705298", "EMERGENCYCID=") in new stack -- Executing [s at macro-outbound-callerid:8] Set("SIP/75002-b7705298", "TRUNKOUTCID=s2") in new stack -- Executing [s at macro-outbound-callerid:9] GotoIf("SIP/75002-b7705298", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s at macro-outbound-callerid:12] ExecIf("SIP/75002-b7705298", "1|Set|CALLERID(all)=s2") in new stack -- Executing [s at macro-outbound-callerid:13] ExecIf("SIP/75002-b7705298", "0|Set|CALLERID(all)=") in new stack -- Executing [s at macro-outbound-callerid:14] ExecIf("SIP/75002-b7705298", "0|SetCallerPres|prohib_passed_screen") in new stack -- Executing [s at macro-dialout-trunk:12] ExecIf("SIP/75002-b7705298", "0|AGI|fixlocalprefix") in new stack -- Executing [s at macro-dialout-trunk:13] Set("SIP/75002-b7705298", "OUTNUM=015921256331") in new stack -- Executing [s at macro-dialout-trunk:14] Set("SIP/75002-b7705298", "custom=SIP/s2") in new stack -- Executing [s at macro-dialout-trunk:15] ExecIf("SIP/75002-b7705298", "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack -- Executing [s at macro-dialout-trunk:16] Macro("SIP/75002-b7705298", "dialout-trunk-predial-hook|") in new stack -- Executing [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/75002-b7705298", "") in new stack -- Executing [s at macro-dialout-trunk:17] GotoIf("SIP/75002-b7705298", "0?bypass|1") in new stack -- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/75002-b7705298", "0?customtrunk") in new stack -- Executing [s at macro-dialout-trunk:19] Dial("SIP/75002-b7705298", "SIP/s2/015921256331|300|M(setmusic^none)Tt") in new stack Audio is at 219.235.234.238 port 17136 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 222.46.18.52:5060: INVITE sip:015921256331 at 222.46.18.52 SIP/2.0 Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport From: "s2" <sip:Unknown at 222.46.18.52>;tag=as75543a2d To: <sip:015921256331 at 222.46.18.52> Contact: <sip:Unknown at 219.235.234.238> Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 26 Mar 2010 06:16:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 3145 3145 IN IP4 219.235.234.238 s=session c=IN IP4 219.235.234.238 t=0 0 m=audio 17136 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called s2/015921256331 gd-branch*CLI> <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;received=58.247.12.18;rport=11028 From: "s2" <sip:Unknown at 222.46.18.52>;tag=as75543a2d To: <sip:015921256331 at 222.46.18.52> Contact: <sip:015921256331 at 222.46.18.52:5060> Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52 CSeq: 102 INVITE Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: timer Server: VOS2009 V2.1.1.8 <-------------> --- (11 headers 0 lines) --- Transmitting (NAT) to 222.46.18.52:5060: ACK sip:015921256331 at 222.46.18.52 SIP/2.0 Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport From: "s2" <sip:Unknown at 222.46.18.52>;tag=as75543a2d To: <sip:015921256331 at 222.46.18.52> Contact: <sip:Unknown at 219.235.234.238> Call-ID: 5cf71e106209cf65344e24031354fbda at 222.46.18.52 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/s2-088f72e8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s at macro-dialout-trunk:20] Goto("SIP/75002-b7705298", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION at macro-dialout-trunk:1] GotoIf("SIP/75002-b7705298", "1?noreport") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,3) -- Executing [s-CONGESTION at macro-dialout-trunk:3] NoOp("SIP/75002-b7705298", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack -- Executing [015921256331 at from-internal:6] Macro("SIP/75002-b7705298", "outisbusy|") in new stack -- Executing [s at macro-outisbusy:1] Playback("SIP/75002-b7705298", "all-circuits-busy-now|noanswer") in new stack -- <SIP/75002-b7705298> Playing 'all-circuits-busy-now' (language 'en') Really destroying SIP dialog '5cf71e106209cf65344e24031354fbda at 222.46.18.52' Method: INVITE -- Executing [s at macro-outisbusy:2] Playback("SIP/75002-b7705298", "pls-try-call-later|noanswer") in new stack -- <SIP/75002-b7705298> Playing 'pls-try-call-later' (language 'en') -- Executing [s at macro-outisbusy:3] Macro("SIP/75002-b7705298", "hangupcall") in new stack -- Executing [s at macro-hangupcall:1] GotoIf("SIP/75002-b7705298", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s at macro-hangupcall:4] GotoIf("SIP/75002-b7705298", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s at macro-hangupcall:7] GotoIf("SIP/75002-b7705298", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s at macro-hangupcall:9] Hangup("SIP/75002-b7705298", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/75002-b7705298' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/75002-b7705298' in macro 'outisbusy' == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/75002-b7705298' == End MixMonitor Recording SIP/75002-b7705298 -- Best Regards! Aaron Chen ------------------------------------------------------------------------------ -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100328/2d5dd5ef/attachment.htm