Hi, I have 2 systems (A and B). I have an 800 number... when someone calls the 800 number it goes: IAX2-->A---IAX---B--->SIP PHONE However.. if the user calling the 800 number is a SIP user that is registered to A it goes: SIP--->A---IAX---B--->SIP PHONE This is the problem... when a call comes in from the IAX2 800 provider, things work fine... however if a SIP user registered to server A dials the 800 number I have it go directly to server B with: extensions_custom.conf:exten => 18666111111,1,Dial(IAX2/callcenter/8666111111) This is what I get in the logs for the failed call: Dec 4 10:01:41 VERBOSE[14045] logger.c: -- Called 126 Dec 4 10:01:41 VERBOSE[14008] logger.c: -- Agent/9999 is ringing Dec 4 10:01:43 DEBUG[15071] chan_sip.c: Auto destroying call 'af65894e0e2728232031b9cf3bf39110@172.16.1.147' Dec 4 10:01:48 DEBUG[15071] chan_sip.c: Auto destroying call '570b281a1e1a00776e19975b59d8764f@172.16.1.120' Dec 4 10:01:48 WARNING[15071] chan_sip.c: Maximum retries exceeded on transmission 1307229403c7ba351483327949372537@63.174.244.175 for seqno 102 (Critical Request) Dec 4 10:01:48 WARNING[15071] chan_sip.c: Hanging up call 1307229403c7ba351483327949372537@63.174.244.175 - no reply to our critical packet. Dec 4 10:01:48 DEBUG[14045] chan_sip.c: update_call_counter(126) - decrement call limit counter Dec 4 10:01:48 VERBOSE[14045] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 4 10:01:48 DEBUG[14008] app_queue.c: Dunno what to do with control type -1 Dec 4 10:01:48 DEBUG[14045] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Dec 4 10:01:48 DEBUG[14045] pbx.c: Expression result is '1' Dec 4 10:01:48 VERBOSE[14045] logger.c: -- Executing GotoIf("Local/126@default-35d0,2", "1?s-CHANUNAVAIL|1") in new stack Dec 4 10:01:48 VERBOSE[14045] logger.c: -- Goto (macro-exten-vm,s-CHANUNAVAIL,1) Dec 4 10:01:48 VERBOSE[14045] logger.c: -- Executing Congestion("Local/126@default-35d0,2", "") in new stack Dec 4 10:01:48 VERBOSE[14008] logger.c: -- Agent/9999 is circuit-busy Dec 4 10:01:48 DEBUG[14008] chan_agent.c: Hangup called for state Down Dec 4 10:01:48 DEBUG[14008] chan_agent.c: Hungup, howlong is 7, autologoff is 28 Dec 4 10:01:48 DEBUG[14008] app_queue.c: Everyone is busy at this time Dec 4 10:01:48 VERBOSE[14045] logger.c: == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 1) exited non-zero on 'Local/126@default-35d0,2' in macro 'exten-vm' Extention 126 is indeed my phone... and my agent ID is 9999. If I call in from the IAX2 provider from outside the system I get: Dec 4 10:06:43 VERBOSE[14214] logger.c: -- Called 126 Dec 4 10:06:43 VERBOSE[14197] logger.c: -- Agent/9999 is ringing Dec 4 10:06:43 DEBUG[15071] chan_sip.c: Auto destroying call '570b281a1e1a00776e19975b59d8764f@172.16.1.120' Dec 4 10:06:43 DEBUG[15071] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4afdb9655ced38f37587d9c50485114e@63.174.244.175' Request 102: Found Dec 4 10:06:43 DEBUG[15071] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4afdb9655ced38f37587d9c50485114e@63.174.244.175' Request 102: Found Dec 4 10:06:43 DEBUG[15071] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4afdb9655ced38f37587d9c50485114e@63.174.244.175' Request 102: Found Dec 4 10:06:43 VERBOSE[14214] logger.c: -- SIP/126-7d1b is ringing Dec 4 10:06:47 VERBOSE[15071] logger.c: -- Started music on hold, class 'default', on Zap/1-1 Dec 4 10:06:47 DEBUG[15071] channel.c: Scheduling timer at 160 sample intervals Dec 4 10:06:47 DEBUG[14188] channel.c: Generator got voice, switching to phase locked mode Dec 4 10:06:47 DEBUG[14188] channel.c: Scheduling timer at 0 sample intervals Dec 4 10:06:47 DEBUG[15071] chan_sip.c: Stopping retransmission on '610dc50c9daebd7d457a841c39f99190@63.174.244.196' of Response 1228778601: Match Found Dec 4 10:06:47 DEBUG[15071] chan_sip.c: Acked pending invite 102 Dec 4 10:06:47 DEBUG[15071] chan_sip.c: Stopping retransmission on '4afdb9655ced38f37587d9c50485114e@63.174.244.175' of Request 102: Match Found Dec 4 10:06:47 DEBUG[15071] chan_sip.c: build_route: Contact hop: <sip:126@63.174.244.196> Dec 4 10:06:47 DEBUG[15059] channel.c: Avoiding initial deadlock for 'SIP/126-7d1b' Dec 4 10:06:47 VERBOSE[14214] logger.c: -- SIP/126-7d1b answered Local/126@default-3489,2 Dec 4 10:06:47 DEBUG[14197] app_queue.c: Dunno what to do with control type -1 Dec 4 10:06:47 VERBOSE[14197] logger.c: -- Agent/9999 answered IAX2/serverA-1 Dec 4 10:06:47 DEBUG[15059] channel.c: Avoiding initial deadlock for 'Local/126@default-3489,2' Dec 4 10:06:47 DEBUG[14197] channel.c: Scheduling timer at 160 sample intervals Dec 4 10:06:47 VERBOSE[14197] logger.c: -- Playing 'custom/CountDown' (language 'en') Dec 4 10:06:48 DEBUG[14214] channel.c: Planning to masquerade channel SIP/126-7d1b into the structure of Local/126@default-3489,1 Dec 4 10:06:48 DEBUG[14214] channel.c: Done planning to masquerade channel SIP/126-7d1b into the structure of Local/126@default-3489,1 Dec 4 10:06:48 DEBUG[14214] chan_local.c: Not posting to queue since already masked on 'Local/126@default-3489,2' Dec 4 10:06:48 DEBUG[14197] channel.c: Got clone lock for masquerade on 'SIP/126-7d1b' at 0x8cb6554 Dec 4 10:06:48 DEBUG[14197] channel.c: Putting channel SIP/126-7d1b in 4/4 formats Dec 4 10:06:48 DEBUG[14197] channel.c: Released clone lock on 'Local/126@default-3489,1<ZOMBIE>' Dec 4 10:06:48 DEBUG[14197] channel.c: Done Masquerading SIP/126-7d1b (6) Dec 4 10:06:48 DEBUG[14197] chan_agent.c: Bridge on 'SIP/126-7d1b' being set to 'Agent/9999' (3) Dec 4 10:06:48 DEBUG[14214] channel.c: Didn't get a frame from channel: Local/126@default-3489,1<ZOMBIE> Dec 4 10:06:48 DEBUG[14214] channel.c: Bridge stops bridging channels Local/126@default-3489,2 and Local/126@default-3489,1<ZOMBIE> Dec 4 10:06:48 DEBUG[14214] app_dial.c: Exiting with DIALSTATUS=ANSWER. What the heck? Anyone have an explination?
Debug of the sip peer 126 shows: -- Called 126 -- Agent/9999 is ringing Retransmitting #1 (NAT) to 63.174.244.196:5060: INVITE sip:126@63.174.244.196 SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 To: <sip:126@63.174.244.196> Contact: <sip:5706016716@63.174.244.175> Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (NAT) to 63.174.244.196:5060: INVITE sip:126@63.174.244.196 SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 To: <sip:126@63.174.244.196> Contact: <sip:5706016716@63.174.244.175> Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to 63.174.244.196:5060: INVITE sip:126@63.174.244.196 SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 To: <sip:126@63.174.244.196> Contact: <sip:5706016716@63.174.244.175> Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #4 (NAT) to 63.174.244.196:5060: INVITE sip:126@63.174.244.196 SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 To: <sip:126@63.174.244.196> Contact: <sip:5706016716@63.174.244.175> Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #5 (NAT) to 63.174.244.196:5060: INVITE sip:126@63.174.244.196 SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 To: <sip:126@63.174.244.196> Contact: <sip:5706016716@63.174.244.175> Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - What is doing this?
Could you explain which devices have what IP and what is behind NAT between what? On 12/4/06, Matt <mhoppes@gmail.com> wrote:> > Debug of the sip peer 126 shows: > > -- Called 126 > -- Agent/9999 is ringing > Retransmitting #1 (NAT) to 63.174.244.196:5060: > INVITE sip:126@63.174.244.196 SIP/2.0 > Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport > From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 > To: <sip:126@63.174.244.196> > Contact: <sip:5706016716@63.174.244.175> > Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 04 Dec 2006 20:42:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 275 > > v=0 > o=root 3555 3555 IN IP4 63.174.244.175 > s=session > c=IN IP4 63.174.244.175 > t=0 0 > m=audio 19720 RTP/AVP 0 97 111 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:111 G726-32/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > Retransmitting #2 (NAT) to 63.174.244.196:5060: > INVITE sip:126@63.174.244.196 SIP/2.0 > Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport > From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 > To: <sip:126@63.174.244.196> > Contact: <sip:5706016716@63.174.244.175> > Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 04 Dec 2006 20:42:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 275 > > v=0 > o=root 3555 3555 IN IP4 63.174.244.175 > s=session > c=IN IP4 63.174.244.175 > t=0 0 > m=audio 19720 RTP/AVP 0 97 111 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:111 G726-32/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > Retransmitting #3 (NAT) to 63.174.244.196:5060: > INVITE sip:126@63.174.244.196 SIP/2.0 > Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport > From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 > To: <sip:126@63.174.244.196> > Contact: <sip:5706016716@63.174.244.175> > Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 04 Dec 2006 20:42:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 275 > > v=0 > o=root 3555 3555 IN IP4 63.174.244.175 > s=session > c=IN IP4 63.174.244.175 > t=0 0 > m=audio 19720 RTP/AVP 0 97 111 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:111 G726-32/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > Retransmitting #4 (NAT) to 63.174.244.196:5060: > INVITE sip:126@63.174.244.196 SIP/2.0 > Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport > From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 > To: <sip:126@63.174.244.196> > Contact: <sip:5706016716@63.174.244.175> > Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 04 Dec 2006 20:42:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 275 > > v=0 > o=root 3555 3555 IN IP4 63.174.244.175 > s=session > c=IN IP4 63.174.244.175 > t=0 0 > m=audio 19720 RTP/AVP 0 97 111 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:111 G726-32/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > Retransmitting #5 (NAT) to 63.174.244.196:5060: > INVITE sip:126@63.174.244.196 SIP/2.0 > Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport > From: "Test VoIP Accounts"" <sip:5706016716@63.174.244.175>;tag=as1a3a38f5 > To: <sip:126@63.174.244.196> > Contact: <sip:5706016716@63.174.244.175> > Call-ID: 1882bae616cf60be14d2436e73c7026a@63.174.244.175 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 04 Dec 2006 20:42:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 275 > > v=0 > o=root 3555 3555 IN IP4 63.174.244.175 > s=session > c=IN IP4 63.174.244.175 > t=0 0 > m=audio 19720 RTP/AVP 0 97 111 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:111 G726-32/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > What is doing this? > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > 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... 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