We have a basic application that runs a SIP channel to pick up a call and process it. We are using Broadvoice and it's been working great. We recently rebooted the machine and now when a call comes in Asterisk picks up the call but does not process it. Asterisk seems to send the call back to Broadvoice. Nothing at all has been changed in the configuration to warrant this. Below is the output of sip debug. Any help would be a life saver!
<-- SIP read from 147.135.20.128:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: "Brooklyn NY"<sip:[EMAIL PROTECTED];user=phone>;tag=ikmn To: "Michael Stearne"<sip:[EMAIL PROTECTED];user=phone> Via: SIP/2.0/UDP 147.135.20.128:5060 Contact: <sip:[EMAIL PROTECTED]:5060> Supported: 100rel RPID-Privacy: party=calling;id-type=subscriber;privacy=off Remote-Party-ID: <sip:[EMAIL PROTECTED]>;screen=yes;party=calling;privacy=off Content-Length: 273 Content-Type: application/sdp v=0 o=2475103479 10 10 IN IP4 147.135.20.247 s=- c=IN IP4 147.135.20.250 t=0 0 m=audio 10690 RTP/AVP 0 8 2 18 96 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:101 telephone-event/8000 --- (12 headers 12 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.135.20.128 : 5060 (non-NAT) Found peer 'sip2.broadvoice.com' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 101 Peer audio RTP is at port 147.135.20.250:10690 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 6092991xxx in from-broadvoice Reliably Transmitting (no NAT) to 147.135.20.128:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 147.135.20.128:5060 From: "Brooklyn NY"<sip:[EMAIL PROTECTED];user=phone>;tag=ikmn To: "Michael Stearne"<sip:[EMAIL PROTECTED];user=phone>;tag=as38d08027 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- <-- SIP read from 147.135.20.128:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: "Brooklyn NY"<sip:[EMAIL PROTECTED];user=phone>;tag=ikmn To: "Michael Stearne"<sip:[EMAIL PROTECTED];user=phone>;tag=as38d08027 Via: SIP/2.0/UDP 147.135.20.128:5060;received=209.3.28.xx Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
