Re: [asterisk-users] Asterisk SIP with no RTP audio port (was Internode weirdness)
On 01/22/11 22:04, Da Rock wrote: On 01/22/11 20:00, Da Rock wrote: On 01/21/11 20:28, Da Rock wrote: On 01/21/11 03:19, Tom Rymes wrote: On 01/19/2011 10:34 PM, Da Rock wrote: WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up. Have you tried disallowing re-invites? Sorry for the delay, but I've tried both yes and no- one of the first things I tried, but I get your reasoning. Thanks Some more information has come to light- bit of luck this clue happen to come to my attention: My provider could be using a Broadworks system. Does that change things much? In my sip debug for the peer it flashed up realm=Broadworks from the peer. Being very new to asterisk and SIP I'm still trying to learn the protocol. Perhaps someone here may be able to correct my understanding if necessary (and a point in the right direction would help significantly). What I wasn't realising was that if I set sip debug on it output the entire sip message. So my output looks like this: -- Executing [0871271201@users:1] Goto(SIP/local ata-0017, internode-outgoing,0871271201,1) in new stack -- Goto (internode-outgoing,0871271201,1) -- Executing [0871271201@internode-outgoing:1] Dial(SIP/local ata-0017, SIP/0871271201@sip-out) in new stack Audio is at 5060 Video is at asterisk ip:5060 Text is at asterisk ip:5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x1000 (g722) to SDP Adding codec 0x8000 (slin16) to SDP Adding video codec 0x10 (h263p) to SDP Adding text codec 0x400 (red) to SDP Adding text codec 0x800 (t140) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 203.2.134.1:5060: INVITE sip:0871271...@sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP asterisk ip:5060;branch=z9hG4bK41104eea Max-Forwards: 70 From: Skinner's Home sip:local ata@asterisk ip;tag=as6683ffea To: sip:0871271...@sip.internode.on.net Contact: sip:local ata@asterisk ip:5060 Call-ID: 5cf55909146ff03907fcc86437809bcc@asterisk ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.1.1 Date: Sat, 22 Jan 2011 11:23:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 736 v=0 o=root 189870721 189870721 IN IP4 asterisk ip s=Asterisk PBX 1.8.1.1 c=IN IP4 asterisk ip b=CT:384 t=0 0 m=audio 17220 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=-- Called 0871271201@sip-out --- SIP read from UDP:203.2.134.1:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk ip:5060;received=my ext ip;branch=z9hG4bK41104eea;rport=61533 From: local ata sip:local ata@asterisk ip;tag=as6683ffea To: sip:0871271...@sip.internode.on.net Call-ID: 5cf55909146ff03907fcc86437809bcc@asterisk ip:5060 CSeq: 102 INVITE - --- (6 headers 0 lines) --- Reliably Transmitting (no NAT) to 203.2.134.1:5060: OPTIONS sip:sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP asterisk ip:5060;branch=z9hG4bK7ab229d7 Max-Forwards: 70 From: Unknown sip:Unknown@asterisk ip;tag=as72e93b63 To: sip:sip.internode.on.net Contact: sip:Unknown@asterisk ip:5060 Call-ID: 068700ad12b1d48a5059a1837a04a8b3@asterisk ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.1.1 Date: Sat, 22 Jan 2011 11:24:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:203.2.134.1:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk ip:5060;received=my ext ip;branch=z9hG4bK7ab229d7;rport=61533 From: Unknown sip:Unknown@asterisk ip;tag=as72e93b63 To: sip:sip.internode.on.net;tag=488684762-1295695493625 Call-ID: 068700ad12b1d48a5059a1837a04a8b3@asterisk ip:5060 CSeq: 102 OPTIONS Allow-Events: call-info,line-seize,dialog,message-summary,as-feature-event Content-Length: 0 - --- (8 headers 0 lines) --- Really destroying SIP dialog '068700ad12b1d48a5059a1837a04a8b3@asterisk ip:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 203.2.134.1:5060: OPTIONS sip:sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP asterisk ip:5060;branch=z9hG4bK53bdf76d Max-Forwards: 70 From: Unknown sip:Unknown@asterisk ip;tag=as2a328631 To: sip:sip.internode.on.net Contact: sip:Unknown@asterisk ip:5060 Call-ID: 42d9681e4b8dcad66879b54c697d7e32@asterisk ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.1.1 Date: Sat, 22 Jan 2011 11:24:00 GMT Allow:
Re: [asterisk-users] Asterisk SIP with no RTP audio port (was Internode weirdness)
On 01/23/11 10:18, Da Rock wrote: On 01/22/11 22:04, Da Rock wrote: On 01/22/11 20:00, Da Rock wrote: On 01/21/11 20:28, Da Rock wrote: On 01/21/11 03:19, Tom Rymes wrote: On 01/19/2011 10:34 PM, Da Rock wrote: WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up. Have you tried disallowing re-invites? Sorry for the delay, but I've tried both yes and no- one of the first things I tried, but I get your reasoning. Thanks Some more information has come to light- bit of luck this clue happen to come to my attention: My provider could be using a Broadworks system. Does that change things much? In my sip debug for the peer it flashed up realm=Broadworks from the peer. Being very new to asterisk and SIP I'm still trying to learn the protocol. Perhaps someone here may be able to correct my understanding if necessary (and a point in the right direction would help significantly). What I wasn't realising was that if I set sip debug on it output the entire sip message. So my output looks like this: -- Executing [0871271201@users:1] Goto(SIP/local ata-0017, internode-outgoing,0871271201,1) in new stack -- Goto (internode-outgoing,0871271201,1) -- Executing [0871271201@internode-outgoing:1] Dial(SIP/local ata-0017, SIP/0871271201@sip-out) in new stack Audio is at 5060 Video is at asterisk ip:5060 Text is at asterisk ip:5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x1000 (g722) to SDP Adding codec 0x8000 (slin16) to SDP Adding video codec 0x10 (h263p) to SDP Adding text codec 0x400 (red) to SDP Adding text codec 0x800 (t140) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 203.2.134.1:5060: INVITE sip:0871271...@sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP asterisk ip:5060;branch=z9hG4bK41104eea Max-Forwards: 70 From: Skinner's Home sip:local ata@asterisk ip;tag=as6683ffea To: sip:0871271...@sip.internode.on.net Contact: sip:local ata@asterisk ip:5060 Call-ID: 5cf55909146ff03907fcc86437809bcc@asterisk ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.1.1 Date: Sat, 22 Jan 2011 11:23:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 736 v=0 o=root 189870721 189870721 IN IP4 asterisk ip s=Asterisk PBX 1.8.1.1 c=IN IP4 asterisk ip b=CT:384 t=0 0 m=audio 17220 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=-- Called 0871271201@sip-out --- SIP read from UDP:203.2.134.1:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk ip:5060;received=my ext ip;branch=z9hG4bK41104eea;rport=61533 From: local ata sip:local ata@asterisk ip;tag=as6683ffea To: sip:0871271...@sip.internode.on.net Call-ID: 5cf55909146ff03907fcc86437809bcc@asterisk ip:5060 CSeq: 102 INVITE - --- (6 headers 0 lines) --- Reliably Transmitting (no NAT) to 203.2.134.1:5060: OPTIONS sip:sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP asterisk ip:5060;branch=z9hG4bK7ab229d7 Max-Forwards: 70 From: Unknown sip:Unknown@asterisk ip;tag=as72e93b63 To: sip:sip.internode.on.net Contact: sip:Unknown@asterisk ip:5060 Call-ID: 068700ad12b1d48a5059a1837a04a8b3@asterisk ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.1.1 Date: Sat, 22 Jan 2011 11:24:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:203.2.134.1:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk ip:5060;received=my ext ip;branch=z9hG4bK7ab229d7;rport=61533 From: Unknown sip:Unknown@asterisk ip;tag=as72e93b63 To: sip:sip.internode.on.net;tag=488684762-1295695493625 Call-ID: 068700ad12b1d48a5059a1837a04a8b3@asterisk ip:5060 CSeq: 102 OPTIONS Allow-Events: call-info,line-seize,dialog,message-summary,as-feature-event Content-Length: 0 - --- (8 headers 0 lines) --- Really destroying SIP dialog '068700ad12b1d48a5059a1837a04a8b3@asterisk ip:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 203.2.134.1:5060: OPTIONS sip:sip.internode.on.net SIP/2.0 Via: SIP/2.0/UDP asterisk ip:5060;branch=z9hG4bK53bdf76d Max-Forwards: 70 From: Unknown sip:Unknown@asterisk ip;tag=as2a328631 To: sip:sip.internode.on.net Contact: sip:Unknown@asterisk ip:5060 Call-ID: 42d9681e4b8dcad66879b54c697d7e32@asterisk ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.1.1 Date:
