Hello all, We have a pathetic legacy PBX that produces the most terrible SIP INVITE packet. In the past we have found a phone that can hope and just used that. We now want to connect the legacy PBX to asterisk, and we're (well, I'm) having problems.
This is the INVITE that's sent to the asterisk server (ip 192.168.0.240) ----------------- INVITE sip:PBX.400T-portal SIP/2.0 To: "01000"<:[EMAIL PROTECTED]> From: <:>;tag=8af2812a Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE Contact: "PBX.400T-portal"<sip:192.168.0.181:5060> Max-Forwards: 70 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Content-Type: application/sdp Date: Sun, 16 Sep 2076 19:04:20 GMT upported: sip-cc, sip-cc-01, timer, replaces User-Agent: PBX.400T-portal Content-Length: 276 ------------- Nasty, eh? At the moment asterisk just says it's not a SIP address and sends a 404. I have put the full trace of how asterisk respnds. Does anyone have any ideas on how to get asterisk to accept an INVITE like this? Thanks, Tom ------ The full asterisk response to the INVITE: ------------- INVITE sip:PBX.400T-portal SIP/2.0 To: "01000"<:[EMAIL PROTECTED]> From: <:>;tag=8af2812a Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE Contact: "PBX.400T-portal"<sip:192.168.0.181:5060> Max-Forwards: 70 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Content-Type: application/sdp Date: Sun, 16 Sep 2076 19:04:20 GMT upported: sip-cc, sip-cc-01, timer, replaces User-Agent: PBX.400T-portal Content-Length: 276 v=0 o=gsm-sip-portal 1 1 IN IP4 192.168.0.181 s=gsm-sip-voice-call c=IN IP4 192.168.0.181 t=0 0 m=audio 8000 RTP/AVP 3 0 8 13 101 a=fmtp:101 0-16 a=rtpmap:3 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 12 lines)--- Using INVITE request as basis request - GSM-SIPCall-Number-1 Sending to 192.168.0.181 : 5060 (NAT) Aug 16 11:58:58 NOTICE[4587]: chan_sip.c:7112 check_user_full: From address missing 'sip:', using it anyway Found peer '01000' Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 13 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.181:8000 Found description format PCMA Found description format PCMU Found description format PCMA Found description format CN Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Aug 16 11:58:58 WARNING[4587]: chan_sip.c:6650 get_destination: Huh? Not a SIP header (:)? Reliably Transmitting (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: <:>;tag=8af2812a To: "01000"<:[EMAIL PROTECTED]>;tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- Retransmitting #1 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: <:>;tag=8af2812a To: "01000"<:[EMAIL PROTECTED]>;tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- Retransmitting #2 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: <:>;tag=8af2812a To: "01000"<:[EMAIL PROTECTED]>;tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- Retransmitting #3 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: <:>;tag=8af2812a To: "01000"<:[EMAIL PROTECTED]>;tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- Retransmitting #4 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: <:>;tag=8af2812a To: "01000"<:[EMAIL PROTECTED]>;tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- Retransmitting #5 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: <:>;tag=8af2812a To: "01000"<:[EMAIL PROTECTED]>;tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- Retransmitting #6 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: <:>;tag=8af2812a To: "01000"<:[EMAIL PROTECTED]>;tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- Aug 16 11:59:18 WARNING[4587]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission GSM-SIPCall-Number-1 for seqno 1 (Critical Response) Destroying call 'GSM-SIPCall-Number-1' _______________________________________________ --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
