[asterisk-users] send a call from A to B use sip trunk prablem
Aaron chen
evane1890 at gmail.com
Thu Mar 25 21:37:30 CDT 2010
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!
Aaron Chen
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