Using Asterisk Management Portal, this config used to work just fine, but it randomly stopped working a few weeks ago. sip show registry shows the number is registering correctly with Broadvoice, & sip debug shows calls coming in but they always get a busy signal. Any idea what's going on? Here's a sip debug:
Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: "XXXX"<sip:[EMAIL PROTECTED];user=phone>;tag=xz13 To: "XXXX"<sip:[EMAIL PROTECTED];user=phone> Via: SIP/2.0/UDP 147.135.12.128:5060 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=2475101431 10 10 IN IP4 147.135.12.247 s=- c=IN IP4 147.135.12.250 t=0 0 m=audio 18092 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 latest request as basis request Sending to 147.135.12.128 : 5060 (non-NAT) 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 147.135.12.250:18092 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 - 0xc (ulaw|alaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found peer 'sip.broadvoice.com' Looking for XXXX in from-pstn Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 147.135.12.128:5060 From: "XXXX"<sip:[EMAIL PROTECTED];user=phone>;tag=xz13 To: "XXXX"<sip:[EMAIL PROTECTED];user=phone>;tag=as54c1e248 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 147.135.12.128:5060 asterisk1*CLI> _______________________________________________ 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
