Aaron chen
2010-Mar-26 02:37 UTC
[asterisk-users] send a call from A to B use sip trunk prablem
i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong?> > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport > From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at 192.168.0.151> > >;tag=as72a55960 > To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>> > Contact: <sip:50005 at 192.168.0.176 <sip%3A50005 at 192.168.0.176>> > Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Fri, 26 Mar 2010 02:12:07 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces > Content-Type: application/sdp > Content-Length: 380 > v=0 > o=root 15081 15081 IN IP4 192.168.0.176 > s=session > c=IN IP4 192.168.0.176 > t=0 0 > m=audio 12726 RTP/AVP 0 18 8 3 4 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:4 G723/8000 > a=fmtp:4 annexa=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > <-------------> > --- (14 headers 18 lines) --- > Sending to 192.168.0.176 : 5060 (NAT) > Using INVITE request as basis request - > 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 > Found peer 's1' > Found RTP audio format 0 > Found RTP audio format 18 > Found RTP audio format 8 > Found RTP audio format 3 > Found RTP audio format 4 > Found RTP audio format 101 > Peer audio RTP is at port 192.168.0.176:12726 > Found audio description format PCMU for ID 0 > Found audio description format G729 for ID 18 > Found audio description format PCMA for ID 8 > Found audio description format GSM for ID 3 > Found audio description format G723 for ID 4 > Found audio description format telephone-event for ID 101 > Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x10f > (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10f > (g723|gsm|ulaw|alaw|g729) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 192.168.0.176:12726 > Looking for 15921256331 in from-internal (domain 192.168.0.151) > list_route: hop: <sip:50005 at 192.168.0.176 <sip%3A50005 at 192.168.0.176>> > gd-branch*CLI> > <--- Transmitting (NAT) to 192.168.0.176:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.0.176:5060 > ;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060 > From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at 192.168.0.151> > >;tag=as72a55960 > To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>> > Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>> > Content-Length: 0 > > <------------> > -- Executing [15921256331 at from-internal:1] > Set("SIP/192.168.0.151-088e7938", "MOHCLASS=none") in new stack > -- Executing [15921256331 at from-internal:2] > Macro("SIP/192.168.0.151-088e7938", "user-callerid|SKIPTTL|") in new stack > -- Executing [s at macro-user-callerid:1] > Set("SIP/192.168.0.151-088e7938", "AMPUSER=50005") in new stack > -- Executing [s at macro-user-callerid:2] > GotoIf("SIP/192.168.0.151-088e7938", "0?report") in new stack > -- Executing [s at macro-user-callerid:3] > ExecIf("SIP/192.168.0.151-088e7938", "1|Set|REALCALLERIDNUM=50005") in new > stack > -- Executing [s at macro-user-callerid:4] > Set("SIP/192.168.0.151-088e7938", "AMPUSER=") in new stack > -- Executing [s at macro-user-callerid:5] > Set("SIP/192.168.0.151-088e7938", "AMPUSERCIDNAME=") in new stack > -- Executing [s at macro-user-callerid:6] > GotoIf("SIP/192.168.0.151-088e7938", "1?report") in new stack > -- Goto (macro-user-callerid,s,10) > -- Executing [s at macro-user-callerid:10] > GotoIf("SIP/192.168.0.151-088e7938", "1?continue") in new stack > -- Goto (macro-user-callerid,s,19) > -- Executing [s at macro-user-callerid:19] > NoOp("SIP/192.168.0.151-088e7938", "Using CallerID "50005" <50005>") in new > stack > -- Executing [15921256331 at from-internal:3] > Set("SIP/192.168.0.151-088e7938", "_NODEST=") in new stack > -- Executing [15921256331 at from-internal:4] > Macro("SIP/192.168.0.151-088e7938", "record-enable||OUT|") in new stack > -- Executing [s at macro-record-enable:1] > GotoIf("SIP/192.168.0.151-088e7938", "1?check") in new stack > -- Goto (macro-record-enable,s,4) > -- Executing [s at macro-record-enable:4] > AGI("SIP/192.168.0.151-088e7938", > "recordingcheck|20100326-101436|1269569676.20") in new stack > -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck > recordingcheck|20100326-101436|1269569676.20: No AMPUSER db entry for . > Not recording > -- AGI Script recordingcheck completed, returning 0 > -- Executing [s at macro-record-enable:5] > MacroExit("SIP/192.168.0.151-088e7938", "") in new stack > -- Executing [15921256331 at from-internal:5] > Macro("SIP/192.168.0.151-088e7938", "dialout-trunk|1|15921256331||") in new > stack > -- Executing [s at macro-dialout-trunk:1] > Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK=1") in new stack > -- Executing [s at macro-dialout-trunk:2] > GosubIf("SIP/192.168.0.151-088e7938", "0?sub-pincheck|s|1") in new stack > -- Executing [s at macro-dialout-trunk:3] > GotoIf("SIP/192.168.0.151-088e7938", "0?disabletrunk|1") in new stack > -- Executing [s at macro-dialout-trunk:4] > Set("SIP/192.168.0.151-088e7938", "DIAL_NUMBER=15921256331") in new stack > -- Executing [s at macro-dialout-trunk:5] > Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Ttr") in new stack > -- Executing [s at macro-dialout-trunk:6] > Set("SIP/192.168.0.151-088e7938", "OUTBOUND_GROUP=OUT_1") in new stack > -- Executing [s at macro-dialout-trunk:7] > GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in new stack > -- Goto (macro-dialout-trunk,s,9) > -- Executing [s at macro-dialout-trunk:9] > GotoIf("SIP/192.168.0.151-088e7938", "0?skipoutcid") in new stack > -- Executing [s at macro-dialout-trunk:10] > Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Tt") in new stack > -- Executing [s at macro-dialout-trunk:11] > Macro("SIP/192.168.0.151-088e7938", "outbound-callerid|1") in new stack > -- Executing [s at macro-outbound-callerid:1] > ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|") in new stack > -- Executing [s at macro-outbound-callerid:2] > ExecIf("SIP/192.168.0.151-088e7938", "0|Set|REALCALLERIDNUM=50005") in new > stack > -- Executing [s at macro-outbound-callerid:3] > GotoIf("SIP/192.168.0.151-088e7938", "1?normcid") in new stack > -- Goto (macro-outbound-callerid,s,6) > -- Executing [s at macro-outbound-callerid:6] > Set("SIP/192.168.0.151-088e7938", "USEROUTCID=") in new stack > -- Executing [s at macro-outbound-callerid:7] > Set("SIP/192.168.0.151-088e7938", "EMERGENCYCID=") in new stack > -- Executing [s at macro-outbound-callerid:8] > Set("SIP/192.168.0.151-088e7938", "TRUNKOUTCID=64858162") in new stack > -- Executing [s at macro-outbound-callerid:9] > GotoIf("SIP/192.168.0.151-088e7938", "1?trunkcid") in new stack > -- Goto (macro-outbound-callerid,s,12) > -- Executing [s at macro-outbound-callerid:12] > ExecIf("SIP/192.168.0.151-088e7938", "1|Set|CALLERID(all)=64858162") in new > stack > -- Executing [s at macro-outbound-callerid:13] > ExecIf("SIP/192.168.0.151-088e7938", "0|Set|CALLERID(all)=") in new stack > -- Executing [s at macro-outbound-callerid:14] > ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|prohib_passed_screen") > in new stack > -- Executing [s at macro-dialout-trunk:12] > ExecIf("SIP/192.168.0.151-088e7938", "0|AGI|fixlocalprefix") in new stack > -- Executing [s at macro-dialout-trunk:13] > Set("SIP/192.168.0.151-088e7938", "OUTNUM=15921256331") in new stack > -- Executing [s at macro-dialout-trunk:14] > Set("SIP/192.168.0.151-088e7938", "custom=ZAP/g0") in new stack > -- Executing [s at macro-dialout-trunk:15] > ExecIf("SIP/192.168.0.151-088e7938", > "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack > -- Executing [s at macro-dialout-trunk:16] > Macro("SIP/192.168.0.151-088e7938", "dialout-trunk-predial-hook|") in new > stack > -- Executing [s at macro-dialout-trunk-predial-hook:1] > MacroExit("SIP/192.168.0.151-088e7938", "") in new stack > -- Executing [s at macro-dialout-trunk:17] > GotoIf("SIP/192.168.0.151-088e7938", "0?bypass|1") in new stack > -- Executing [s at macro-dialout-trunk:18] > GotoIf("SIP/192.168.0.151-088e7938", "0?customtrunk") in new stack > -- Executing [s at macro-dialout-trunk:19] > Dial("SIP/192.168.0.151-088e7938", > "ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [s at macro-dialout-trunk:20] > Goto("SIP/192.168.0.151-088e7938", "s-CHANUNAVAIL|1") in new stack > -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) > -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:1] > GotoIf("SIP/192.168.0.151-088e7938", "1?noreport") in new stack > -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) > -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:3] > NoOp("SIP/192.168.0.151-088e7938", "TRUNK Dial failed due to CHANUNAVAIL > (hangupcause: 58) - failing through to other trunks") in new stack > -- Executing [15921256331 at from-internal:6] > Macro("SIP/192.168.0.151-088e7938", "outisbusy|") in new stack > -- Executing [s at macro-outisbusy:1] > Playback("SIP/192.168.0.151-088e7938", "all-circuits-busy-now|noanswer") in > new stack > -- Executing [s at macro-outisbusy:2] > Playback("SIP/192.168.0.151-088e7938", "pls-try-call-later|noanswer") in new > stack > -- Executing [s at macro-outisbusy:3] Macro("SIP/192.168.0.151-088e7938", > "hangupcall") in new stack > -- Executing [s at macro-hangupcall:1] > GotoIf("SIP/192.168.0.151-088e7938", "1?skiprg") in new stack > -- Goto (macro-hangupcall,s,4) > -- Executing [s at macro-hangupcall:4] > GotoIf("SIP/192.168.0.151-088e7938", "1?skipblkvm") in new stack > -- Goto (macro-hangupcall,s,7) > -- Executing [s at macro-hangupcall:7] > GotoIf("SIP/192.168.0.151-088e7938", "1?theend") in new stack > -- Goto (macro-hangupcall,s,9) > -- Executing [s at macro-hangupcall:9] > Hangup("SIP/192.168.0.151-088e7938", "") in new stack > == Spawn extension (macro-hangupcall, s, 9) exited non-zero on > 'SIP/192.168.0.151-088e7938' in macro 'hangupcall' > == Spawn extension (macro-hangupcall, s, 9) exited non-zero on > 'SIP/192.168.0.151-088e7938' in macro 'outisbusy' > == Spawn extension (macro-hangupcall, s, 9) exited non-zero on > 'SIP/192.168.0.151-088e7938' > Scheduling destruction of SIP dialog > '28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151' in 6400 ms (Method: > INVITE) > gd-branch*CLI> > <--- Reliably Transmitting (NAT) to 192.168.0.176:5060 ---> > SIP/2.0 488 Not Acceptable Here > Via: SIP/2.0/UDP 192.168.0.176:5060 > ;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060 > From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at 192.168.0.151> > >;tag=as72a55960 > To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151> > >;tag=as12db2697 > Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > <------------> > gd-branch*CLI> > <--- SIP read from 192.168.0.176:5060 ---> > ACK sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport > From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at 192.168.0.151> > >;tag=as72a55960 > To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151> > >;tag=as12db2697 > Contact: <sip:50005 at 192.168.0.176 <sip%3A50005 at 192.168.0.176>> > Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > <-------------> > --- (10 headers 0 lines) --- > sip no debug > SIP Debugging Disabled >Best regards! 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Alyed
2010-Mar-26 05:44 UTC
[asterisk-users] send a call from A to B use sip trunk prablem
it doesn't seems to be a problem of communication between A and B> -- Executing [s at macro-dialout-trunk:19]Dial("SIP/192.168.0.151-088e7938", "ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack> == Everyone is busy/congested at this time (1:0/0/1)That's says it's more a problem with your Zap channels than your SIP connection. First try playing a sound in B when receiving the call, that way you can be sure the connection is ok. If that one works then move to PSTN. Alyed 2010/3/25 Aaron chen <evane1890 at gmail.com>> i have a prablom here, > > i want to send a call from A to B use sip trunk , > > the call can sended B,but not work to PSTN. > > the message from B server. help pls,what's rong? > > > >> >> <--- SIP read from 192.168.0.176:5060 ---> >> INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 >> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport >> From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at 192.168.0.151> >> >;tag=as72a55960 >> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>> >> Contact: <sip:50005 at 192.168.0.176 <sip%3A50005 at 192.168.0.176>> >> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Date: Fri, 26 Mar 2010 02:12:07 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO >> Supported: replaces >> Content-Type: application/sdp >> Content-Length: 380 >> v=0 >> o=root 15081 15081 IN IP4 192.168.0.176 >> s=session >> c=IN IP4 192.168.0.176 >> t=0 0 >> m=audio 12726 RTP/AVP 0 18 8 3 4 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:4 G723/8000 >> a=fmtp:4 annexa=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> <-------------> >> --- (14 headers 18 lines) --- >> Sending to 192.168.0.176 : 5060 (NAT) >> Using INVITE request as basis request - >> 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 >> Found peer 's1' >> Found RTP audio format 0 >> Found RTP audio format 18 >> Found RTP audio format 8 >> Found RTP audio format 3 >> Found RTP audio format 4 >> Found RTP audio format 101 >> Peer audio RTP is at port 192.168.0.176:12726 >> Found audio description format PCMU for ID 0 >> Found audio description format G729 for ID 18 >> Found audio description format PCMA for ID 8 >> Found audio description format GSM for ID 3 >> Found audio description format G723 for ID 4 >> Found audio description format telephone-event for ID 101 >> Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x10f >> (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10f >> (g723|gsm|ulaw|alaw|g729) >> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 >> (telephone-event), combined - 0x1 (telephone-event) >> Peer audio RTP is at port 192.168.0.176:12726 >> Looking for 15921256331 in from-internal (domain 192.168.0.151) >> list_route: hop: <sip:50005 at 192.168.0.176 <sip%3A50005 at 192.168.0.176>> >> gd-branch*CLI> >> <--- Transmitting (NAT) to 192.168.0.176:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.0.176:5060 >> ;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060 >> From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at 192.168.0.151> >> >;tag=as72a55960 >> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>> >> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Supported: replaces >> Contact: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151> >> > >> Content-Length: 0 >> >> <------------> >> -- Executing [15921256331 at from-internal:1] >> Set("SIP/192.168.0.151-088e7938", "MOHCLASS=none") in new stack >> -- Executing [15921256331 at from-internal:2] >> Macro("SIP/192.168.0.151-088e7938", "user-callerid|SKIPTTL|") in new stack >> -- Executing [s at macro-user-callerid:1] >> Set("SIP/192.168.0.151-088e7938", "AMPUSER=50005") in new stack >> -- Executing [s at macro-user-callerid:2] >> GotoIf("SIP/192.168.0.151-088e7938", "0?report") in new stack >> -- Executing [s at macro-user-callerid:3] >> ExecIf("SIP/192.168.0.151-088e7938", "1|Set|REALCALLERIDNUM=50005") in new >> stack >> -- Executing [s at macro-user-callerid:4] >> Set("SIP/192.168.0.151-088e7938", "AMPUSER=") in new stack >> -- Executing [s at macro-user-callerid:5] >> Set("SIP/192.168.0.151-088e7938", "AMPUSERCIDNAME=") in new stack >> -- Executing [s at macro-user-callerid:6] >> GotoIf("SIP/192.168.0.151-088e7938", "1?report") in new stack >> -- Goto (macro-user-callerid,s,10) >> -- Executing [s at macro-user-callerid:10] >> GotoIf("SIP/192.168.0.151-088e7938", "1?continue") in new stack >> -- Goto (macro-user-callerid,s,19) >> -- Executing [s at macro-user-callerid:19] >> NoOp("SIP/192.168.0.151-088e7938", "Using CallerID "50005" <50005>") in new >> stack >> -- Executing [15921256331 at from-internal:3] >> Set("SIP/192.168.0.151-088e7938", "_NODEST=") in new stack >> -- Executing [15921256331 at from-internal:4] >> Macro("SIP/192.168.0.151-088e7938", "record-enable||OUT|") in new stack >> -- Executing [s at macro-record-enable:1] >> GotoIf("SIP/192.168.0.151-088e7938", "1?check") in new stack >> -- Goto (macro-record-enable,s,4) >> -- Executing [s at macro-record-enable:4] >> AGI("SIP/192.168.0.151-088e7938", >> "recordingcheck|20100326-101436|1269569676.20") in new stack >> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck >> recordingcheck|20100326-101436|1269569676.20: No AMPUSER db entry for . >> Not recording >> -- AGI Script recordingcheck completed, returning 0 >> -- Executing [s at macro-record-enable:5] >> MacroExit("SIP/192.168.0.151-088e7938", "") in new stack >> -- Executing [15921256331 at from-internal:5] >> Macro("SIP/192.168.0.151-088e7938", "dialout-trunk|1|15921256331||") in new >> stack >> -- Executing [s at macro-dialout-trunk:1] >> Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK=1") in new stack >> -- Executing [s at macro-dialout-trunk:2] >> GosubIf("SIP/192.168.0.151-088e7938", "0?sub-pincheck|s|1") in new stack >> -- Executing [s at macro-dialout-trunk:3] >> GotoIf("SIP/192.168.0.151-088e7938", "0?disabletrunk|1") in new stack >> -- Executing [s at macro-dialout-trunk:4] >> Set("SIP/192.168.0.151-088e7938", "DIAL_NUMBER=15921256331") in new stack >> -- Executing [s at macro-dialout-trunk:5] >> Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Ttr") in new stack >> -- Executing [s at macro-dialout-trunk:6] >> Set("SIP/192.168.0.151-088e7938", "OUTBOUND_GROUP=OUT_1") in new stack >> -- Executing [s at macro-dialout-trunk:7] >> GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in new stack >> -- Goto (macro-dialout-trunk,s,9) >> -- Executing [s at macro-dialout-trunk:9] >> GotoIf("SIP/192.168.0.151-088e7938", "0?skipoutcid") in new stack >> -- Executing [s at macro-dialout-trunk:10] >> Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Tt") in new stack >> -- Executing [s at macro-dialout-trunk:11] >> Macro("SIP/192.168.0.151-088e7938", "outbound-callerid|1") in new stack >> -- Executing [s at macro-outbound-callerid:1] >> ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|") in new stack >> -- Executing [s at macro-outbound-callerid:2] >> ExecIf("SIP/192.168.0.151-088e7938", "0|Set|REALCALLERIDNUM=50005") in new >> stack >> -- Executing [s at macro-outbound-callerid:3] >> GotoIf("SIP/192.168.0.151-088e7938", "1?normcid") in new stack >> -- Goto (macro-outbound-callerid,s,6) >> -- Executing [s at macro-outbound-callerid:6] >> Set("SIP/192.168.0.151-088e7938", "USEROUTCID=") in new stack >> -- Executing [s at macro-outbound-callerid:7] >> Set("SIP/192.168.0.151-088e7938", "EMERGENCYCID=") in new stack >> -- Executing [s at macro-outbound-callerid:8] >> Set("SIP/192.168.0.151-088e7938", "TRUNKOUTCID=64858162") in new stack >> -- Executing [s at macro-outbound-callerid:9] >> GotoIf("SIP/192.168.0.151-088e7938", "1?trunkcid") in new stack >> -- Goto (macro-outbound-callerid,s,12) >> -- Executing [s at macro-outbound-callerid:12] >> ExecIf("SIP/192.168.0.151-088e7938", "1|Set|CALLERID(all)=64858162") in new >> stack >> -- Executing [s at macro-outbound-callerid:13] >> ExecIf("SIP/192.168.0.151-088e7938", "0|Set|CALLERID(all)=") in new stack >> -- Executing [s at macro-outbound-callerid:14] >> ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|prohib_passed_screen") >> in new stack >> -- Executing [s at macro-dialout-trunk:12] >> ExecIf("SIP/192.168.0.151-088e7938", "0|AGI|fixlocalprefix") in new stack >> -- Executing [s at macro-dialout-trunk:13] >> Set("SIP/192.168.0.151-088e7938", "OUTNUM=15921256331") in new stack >> -- Executing [s at macro-dialout-trunk:14] >> Set("SIP/192.168.0.151-088e7938", "custom=ZAP/g0") in new stack >> -- Executing [s at macro-dialout-trunk:15] >> ExecIf("SIP/192.168.0.151-088e7938", >> "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack >> -- Executing [s at macro-dialout-trunk:16] >> Macro("SIP/192.168.0.151-088e7938", "dialout-trunk-predial-hook|") in new >> stack >> -- Executing [s at macro-dialout-trunk-predial-hook:1] >> MacroExit("SIP/192.168.0.151-088e7938", "") in new stack >> -- Executing [s at macro-dialout-trunk:17] >> GotoIf("SIP/192.168.0.151-088e7938", "0?bypass|1") in new stack >> -- Executing [s at macro-dialout-trunk:18] >> GotoIf("SIP/192.168.0.151-088e7938", "0?customtrunk") in new stack >> -- Executing [s at macro-dialout-trunk:19] >> Dial("SIP/192.168.0.151-088e7938", >> "ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack >> == Everyone is busy/congested at this time (1:0/0/1) >> -- Executing [s at macro-dialout-trunk:20] >> Goto("SIP/192.168.0.151-088e7938", "s-CHANUNAVAIL|1") in new stack >> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) >> -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:1] >> GotoIf("SIP/192.168.0.151-088e7938", "1?noreport") in new stack >> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) >> -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:3] >> NoOp("SIP/192.168.0.151-088e7938", "TRUNK Dial failed due to CHANUNAVAIL >> (hangupcause: 58) - failing through to other trunks") in new stack >> -- Executing [15921256331 at from-internal:6] >> Macro("SIP/192.168.0.151-088e7938", "outisbusy|") in new stack >> -- Executing [s at macro-outisbusy:1] >> Playback("SIP/192.168.0.151-088e7938", "all-circuits-busy-now|noanswer") in >> new stack >> -- Executing [s at macro-outisbusy:2] >> Playback("SIP/192.168.0.151-088e7938", "pls-try-call-later|noanswer") in new >> stack >> -- Executing [s at macro-outisbusy:3] >> Macro("SIP/192.168.0.151-088e7938", "hangupcall") in new stack >> -- Executing [s at macro-hangupcall:1] >> GotoIf("SIP/192.168.0.151-088e7938", "1?skiprg") in new stack >> -- Goto (macro-hangupcall,s,4) >> -- Executing [s at macro-hangupcall:4] >> GotoIf("SIP/192.168.0.151-088e7938", "1?skipblkvm") in new stack >> -- Goto (macro-hangupcall,s,7) >> -- Executing [s at macro-hangupcall:7] >> GotoIf("SIP/192.168.0.151-088e7938", "1?theend") in new stack >> -- Goto (macro-hangupcall,s,9) >> -- Executing [s at macro-hangupcall:9] >> Hangup("SIP/192.168.0.151-088e7938", "") in new stack >> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on >> 'SIP/192.168.0.151-088e7938' in macro 'hangupcall' >> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on >> 'SIP/192.168.0.151-088e7938' in macro 'outisbusy' >> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on >> 'SIP/192.168.0.151-088e7938' >> Scheduling destruction of SIP dialog >> '28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151'<%2728272ebb12ee6e4c1f06fca651456469 at 192.168.0.151%27>in 6400 ms (Method: INVITE) >> gd-branch*CLI> >> <--- Reliably Transmitting (NAT) to 192.168.0.176:5060 ---> >> SIP/2.0 488 Not Acceptable Here >> Via: SIP/2.0/UDP 192.168.0.176:5060 >> ;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060 >> From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at 192.168.0.151> >> >;tag=as72a55960 >> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151> >> >;tag=as12db2697 >> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Supported: replaces >> Content-Length: 0 >> >> <------------> >> gd-branch*CLI> >> <--- SIP read from 192.168.0.176:5060 ---> >> ACK sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 >> Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport >> From: "50005" <sip:50005 at 192.168.0.151 <sip%3A50005 at 192.168.0.151> >> >;tag=as72a55960 >> To: <sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151> >> >;tag=as12db2697 >> Contact: <sip:50005 at 192.168.0.176 <sip%3A50005 at 192.168.0.176>> >> Call-ID: 28272ebb12ee6e4c1f06fca651456469 at 192.168.0.151 >> CSeq: 102 ACK >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> <-------------> >> --- (10 headers 0 lines) --- >> sip no debug >> SIP Debugging Disabled >> > > > 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/20100325/1001355a/attachment.htm