My server use public ip, so no nat issues, here is the out of sip debug:
<-------------> --- (10 headers 0 lines) --- Sending to 213.165.32.100 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 213.165.32.100:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: <sip:9991...@213.165.32.100>;tag=3466008105-77358 To: 966599740196 <sip:966599740...@213.165.32.100>;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> elastix*CLI> <--- Transmitting (no NAT) to 213.165.32.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: <sip:9991...@213.165.32.100>;tag=3466008105-77358 To: 966599740196 <sip:966599740...@213.165.32.100>;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> -- Hungup 'IAX2/99999-4490' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966599740196, 4) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' -- Executing [...@macro-dialout-trunk:1] Macro("SIP/213.165.32.100-b7c10ad8", "hangupcall|") in new stack -- Executing [...@macro-hangupcall:1] ResetCDR("SIP/213.165.32.100-b7c10ad8", "w") in new stack -- Executing [...@macro-hangupcall:2] NoCDR("SIP/213.165.32.100-b7c10ad8", "") in new stack -- Executing [...@macro-hangupcall:3] GotoIf("SIP/213.165.32.100-b7c10ad8", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf("SIP/213.165.32.100-b7c10ad8", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf("SIP/213.165.32.100-b7c10ad8", "1?theend") in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup("SIP/213.165.32.100-b7c10ad8", "") in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' elastix*CLI> thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 1:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying "hanging up call XXXX, no reply to our critical package". see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI> -- Hungup 'IAX2/99999-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on 'SIP/213.165.32.100-b7d21018' -- Executing [...@macro-dialout-trunk:1] Macro("SIP/213.165.32.100-b7d21018", "hangupcall|") in new stack -- Executing [...@macro-hangupcall:1] ResetCDR("SIP/213.165.32.100-b7d21018", "w") in new stack -- Executing [...@macro-hangupcall:2] NoCDR("SIP/213.165.32.100-b7d21018", "") in new stack -- Executing [...@macro-hangupcall:3] GotoIf("SIP/213.165.32.100-b7d21018", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf("SIP/213.165.32.100-b7d21018", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf("SIP/213.165.32.100-b7d21018", "1?theend") in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup("SIP/213.165.32.100-b7d21018", "") in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7d21018' elastix*CLI>
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