Greg Blakely
2004-Aug-04 19:11 UTC
[Asterisk-Users] New Head Appears to Break SIP to iConnect
Folks, I have to admit that I MAY have changed something (at someone's advice) on a previous CVS head (May 28), but I'm not sure. I think that it had to do with changing "digest realm," but that may be a different issue. At any rate, I had both incoming and outgoing with iConnectHere. Now, I made exactly ONE change: I upgraded to the CVS head dated 7/30. I still have outgoing SIP via iconnect, but the incoming just hangs, and finally times out to an iconnect intercept recording (Your Call Cannot be completed). I've done a 'sip debug,' and it appears that I've got the old 407 Error: (209.98.47.209 is me; 213.xxx.xxx.xxx is iconnect) lizzie*CLI> Sip read: INVITE sip:16126053676@209.98.47.209:5060 SIP/2.0 Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183 Via: SIP/2.0/UDP 213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1 Via: SIP/2.0/UDP 213.137.65.239:5060;received=213.137.65.239 To: <sip:16126053676@213.137.73.179> From: <sip:9529496767@213.137.65.239>;tag=31062DE8-21CE Call-ID: 359C5C75-E5BB11D8-9120D756-77435478@213.137.65.239 CSeq: 101 INVITE Contact: <sip:9529496767@213.137.65.239:5060> Record-Route: <sip:16126053676@213.137.73.140:5060;maddr=213.137.73.183> Record-Route: <sip:9529496767.22b27932-ad4142ba-5399fdb0-571ef11f@213.137.65.239:5060; maddr=213.137.73.41> Content-Type: application/sdp Content-Length: 148 v=0 o=CiscoSystemsSIP-GW-UserAgent 5851 2446 IN IP4 213.137.65.239 s=SIP Call c=IN IP4 213.137.65.239 t=0 0 m=audio 18698 RTP/AVP 4 0 8 2 101 13 headers, 6 lines Using latest request as basis request Sending to 213.137.73.140 : 5060 (non-NAT) Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 101 Peer RTP is at port 213.137.65.239:0 Capabilities: us - 0xc(ULAW|ALAW), peer - audio=0x1d(G723|ULAW|ALAW|G726)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'iconnect' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183 Via: SIP/2.0/UDP 213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1 Via: SIP/2.0/UDP 213.137.65.239:5060;received=213.137.65.239 From: <sip:9529496767@213.137.65.239>;tag=31062DE8-21CE To: <sip:16126053676@213.137.73.179>;tag=as702accc9 Call-ID: 359C5C75-E5BB11D8-9120D756-77435478@213.137.65.239 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:16126053676@209.98.47.209> Proxy-Authenticate: Digest realm="asterisk", nonce="60c6600e" Content-Length: 0 to 213.137.73.140:5060 Scheduling destruction of call '359C5C75-E5BB11D8-9120D756-77435478@213.137.65.239' in 15000 ms lizzie*CLI> Sip read: ACK sip:16126053676@209.98.47.209:5060 SIP/2.0 Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183 Via: SIP/2.0/UDP 213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1 From: <sip:9529496767@213.137.65.239>;tag=31062DE8-21CE To: <sip:16126053676@213.137.73.179>;tag=as702accc9 Call-ID: 359C5C75-E5BB11D8-9120D756-77435478@213.137.65.239 CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines lizzie*CLI> [root@lizzie asterisk]#