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:[email protected] <sip%[email protected]>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport > From: "50005" <sip:[email protected] <sip%[email protected]> > >;tag=as72a55960 > To: <sip:[email protected] <sip%[email protected]>> > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > 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 - > [email protected] > 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:[email protected] <sip%[email protected]>> > 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:[email protected] <sip%[email protected]> > >;tag=as72a55960 > To: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected] <sip%[email protected]>> > Content-Length: 0 > > <------------> > -- Executing [15921256...@from-internal:1] > Set("SIP/192.168.0.151-088e7938", "MOHCLASS=none") in new stack > -- Executing [15921256...@from-internal:2] > Macro("SIP/192.168.0.151-088e7938", "user-callerid|SKIPTTL|") in new stack > -- Executing [...@macro-user-callerid:1] > Set("SIP/192.168.0.151-088e7938", "AMPUSER=50005") in new stack > -- Executing [...@macro-user-callerid:2] > GotoIf("SIP/192.168.0.151-088e7938", "0?report") in new stack > -- Executing [...@macro-user-callerid:3] > ExecIf("SIP/192.168.0.151-088e7938", "1|Set|REALCALLERIDNUM=50005") in new > stack > -- Executing [...@macro-user-callerid:4] > Set("SIP/192.168.0.151-088e7938", "AMPUSER=") in new stack > -- Executing [...@macro-user-callerid:5] > Set("SIP/192.168.0.151-088e7938", "AMPUSERCIDNAME=") in new stack > -- Executing [...@macro-user-callerid:6] > GotoIf("SIP/192.168.0.151-088e7938", "1?report") in new stack > -- Goto (macro-user-callerid,s,10) > -- Executing [...@macro-user-callerid:10] > GotoIf("SIP/192.168.0.151-088e7938", "1?continue") in new stack > -- Goto (macro-user-callerid,s,19) > -- Executing [...@macro-user-callerid:19] > NoOp("SIP/192.168.0.151-088e7938", "Using CallerID "50005" <50005>") in new > stack > -- Executing [15921256...@from-internal:3] > Set("SIP/192.168.0.151-088e7938", "_NODEST=") in new stack > -- Executing [15921256...@from-internal:4] > Macro("SIP/192.168.0.151-088e7938", "record-enable||OUT|") in new stack > -- Executing [...@macro-record-enable:1] > GotoIf("SIP/192.168.0.151-088e7938", "1?check") in new stack > -- Goto (macro-record-enable,s,4) > -- Executing [...@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 [...@macro-record-enable:5] > MacroExit("SIP/192.168.0.151-088e7938", "") in new stack > -- Executing [15921256...@from-internal:5] > Macro("SIP/192.168.0.151-088e7938", "dialout-trunk|1|15921256331||") in new > stack > -- Executing [...@macro-dialout-trunk:1] > Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK=1") in new stack > -- Executing [...@macro-dialout-trunk:2] > GosubIf("SIP/192.168.0.151-088e7938", "0?sub-pincheck|s|1") in new stack > -- Executing [...@macro-dialout-trunk:3] > GotoIf("SIP/192.168.0.151-088e7938", "0?disabletrunk|1") in new stack > -- Executing [...@macro-dialout-trunk:4] > Set("SIP/192.168.0.151-088e7938", "DIAL_NUMBER=15921256331") in new stack > -- Executing [...@macro-dialout-trunk:5] > Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Ttr") in new stack > -- Executing [...@macro-dialout-trunk:6] > Set("SIP/192.168.0.151-088e7938", "OUTBOUND_GROUP=OUT_1") in new stack > -- Executing [...@macro-dialout-trunk:7] > GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in new stack > -- Goto (macro-dialout-trunk,s,9) > -- Executing [...@macro-dialout-trunk:9] > GotoIf("SIP/192.168.0.151-088e7938", "0?skipoutcid") in new stack > -- Executing [...@macro-dialout-trunk:10] > Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Tt") in new stack > -- Executing [...@macro-dialout-trunk:11] > Macro("SIP/192.168.0.151-088e7938", "outbound-callerid|1") in new stack > -- Executing [...@macro-outbound-callerid:1] > ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|") in new stack > -- Executing [...@macro-outbound-callerid:2] > ExecIf("SIP/192.168.0.151-088e7938", "0|Set|REALCALLERIDNUM=50005") in new > stack > -- Executing [...@macro-outbound-callerid:3] > GotoIf("SIP/192.168.0.151-088e7938", "1?normcid") in new stack > -- Goto (macro-outbound-callerid,s,6) > -- Executing [...@macro-outbound-callerid:6] > Set("SIP/192.168.0.151-088e7938", "USEROUTCID=") in new stack > -- Executing [...@macro-outbound-callerid:7] > Set("SIP/192.168.0.151-088e7938", "EMERGENCYCID=") in new stack > -- Executing [...@macro-outbound-callerid:8] > Set("SIP/192.168.0.151-088e7938", "TRUNKOUTCID=64858162") in new stack > -- Executing [...@macro-outbound-callerid:9] > GotoIf("SIP/192.168.0.151-088e7938", "1?trunkcid") in new stack > -- Goto (macro-outbound-callerid,s,12) > -- Executing [...@macro-outbound-callerid:12] > ExecIf("SIP/192.168.0.151-088e7938", "1|Set|CALLERID(all)=64858162") in new > stack > -- Executing [...@macro-outbound-callerid:13] > ExecIf("SIP/192.168.0.151-088e7938", "0|Set|CALLERID(all)=") in new stack > -- Executing [...@macro-outbound-callerid:14] > ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|prohib_passed_screen") > in new stack > -- Executing [...@macro-dialout-trunk:12] > ExecIf("SIP/192.168.0.151-088e7938", "0|AGI|fixlocalprefix") in new stack > -- Executing [...@macro-dialout-trunk:13] > Set("SIP/192.168.0.151-088e7938", "OUTNUM=15921256331") in new stack > -- Executing [...@macro-dialout-trunk:14] > Set("SIP/192.168.0.151-088e7938", "custom=ZAP/g0") in new stack > -- Executing [...@macro-dialout-trunk:15] > ExecIf("SIP/192.168.0.151-088e7938", > "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack > -- Executing [...@macro-dialout-trunk:16] > Macro("SIP/192.168.0.151-088e7938", "dialout-trunk-predial-hook|") in new > stack > -- Executing [...@macro-dialout-trunk-predial-hook:1] > MacroExit("SIP/192.168.0.151-088e7938", "") in new stack > -- Executing [...@macro-dialout-trunk:17] > GotoIf("SIP/192.168.0.151-088e7938", "0?bypass|1") in new stack > -- Executing [...@macro-dialout-trunk:18] > GotoIf("SIP/192.168.0.151-088e7938", "0?customtrunk") in new stack > -- Executing [...@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 [...@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-chanunav...@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-chanunav...@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 [15921256...@from-internal:6] > Macro("SIP/192.168.0.151-088e7938", "outisbusy|") in new stack > -- Executing [...@macro-outisbusy:1] > Playback("SIP/192.168.0.151-088e7938", "all-circuits-busy-now|noanswer") in > new stack > -- Executing [...@macro-outisbusy:2] > Playback("SIP/192.168.0.151-088e7938", "pls-try-call-later|noanswer") in new > stack > -- Executing [...@macro-outisbusy:3] Macro("SIP/192.168.0.151-088e7938", > "hangupcall") in new stack > -- Executing [...@macro-hangupcall:1] > GotoIf("SIP/192.168.0.151-088e7938", "1?skiprg") in new stack > -- Goto (macro-hangupcall,s,4) > -- Executing [...@macro-hangupcall:4] > GotoIf("SIP/192.168.0.151-088e7938", "1?skipblkvm") in new stack > -- Goto (macro-hangupcall,s,7) > -- Executing [...@macro-hangupcall:7] > GotoIf("SIP/192.168.0.151-088e7938", "1?theend") in new stack > -- Goto (macro-hangupcall,s,9) > -- Executing [...@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 > '[email protected]' 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:[email protected] <sip%[email protected]> > >;tag=as72a55960 > To: <sip:[email protected] <sip%[email protected]> > >;tag=as12db2697 > Call-ID: [email protected] > 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:[email protected] <sip%[email protected]>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport > From: "50005" <sip:[email protected] <sip%[email protected]> > >;tag=as72a55960 > To: <sip:[email protected] <sip%[email protected]> > >;tag=as12db2697 > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > 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
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