Using AMP, the configuration I have used to work fine with Broadvoice. Now it gets a busy signal every time. I've checked "sip show registry" and it says it's registered just fine. I've tried "sip debug" and it shows calls coming in, but they always get a busy signal & I can't tell why. Here's a SIP Debug output: Sip read: INVITE sip:XXXX@192.168.1.107:5060 SIP/2.0 Call-ID: ff01aa-43@147.135.12.128 CSeq: 1 INVITE From: "XXXX"<sip:XXXX@147.135.12.128;user=phone>;tag=xz13 To: "XXXX"<sip:s@192.168.1.107;user=phone> Via: SIP/2.0/UDP 147.135.12.128:5060 Contact: sip:XXXX@147.135.12.128:5060 Supported: 100rel RPID-Privacy: party=calling;id-type=subscriber;privacy=off Remote-Party-ID: <sip:XXXX@147.135.12.128>;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:XXXX@147.135.12.128;user=phone>;tag=xz13 To: "XXXX"<sip:s@192.168.1.107;user=phone>;tag=as54c1e248 Call-ID: ff01aa-43@147.135.12.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:XXXX@192.168.1.107> Content-Length: 0 to 147.135.12.128:5060 asterisk1*CLI>
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:XXXX@192.168.1.107:5060 SIP/2.0 Call-ID: ff01aa-43@147.135.12.128 CSeq: 1 INVITE From: "XXXX"<sip:XXXX@147.135.12.128;user=phone>;tag=xz13 To: "XXXX"<sip:s@192.168.1.107;user=phone> Via: SIP/2.0/UDP 147.135.12.128:5060 Contact: sip:XXXX@147.135.12.128:5060 Supported: 100rel RPID-Privacy: party=calling;id-type=subscriber;privacy=off Remote-Party-ID: <sip:XXXX@147.135.12.128>;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:XXXX@147.135.12.128;user=phone>;tag=xz13 To: "XXXX"<sip:s@192.168.1.107;user=phone>;tag=as54c1e248 Call-ID: ff01aa-43@147.135.12.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:XXXX@192.168.1.107> Content-Length: 0 to 147.135.12.128:5060 asterisk1*CLI>
Using Asterisk Management Portal with Broadvoice. It used to work just fine; calls would come in and be answered with no trouble at all. A few weeks ago with no configuration changes at all Asterisk stopped picking up calls and started giving a busy signal whenever someone calls. I've tried rebooting the system many times, and "sip show registry" shows it's registering correctly with Broadvoice. Sip debug shows the UDP packets correctly hit the system on port 5060, but the call is rejected\busy instead of answered. Here's a SIP debug or a call coming in and being busy. Any clues? Sip read: INVITE sip:XXXX@192.168.1.107:5060 SIP/2.0 Call-ID: ff01aa-43@147.135.12.128 CSeq: 1 INVITE From: "XXXX"<sip:XXXX@147.135.12.128;user=phone>;tag=xz13 To: "XXXX"<sip:s@192.168.1.107;user=phone> Via: SIP/2.0/UDP 147.135.12.128:5060 Contact: sip:XXXX@147.135.12.128:5060 Supported: 100rel RPID-Privacy: party=calling;id-type=subscriber;privacy=off Remote-Party-ID: <sip:XXXX@147.135.12.128>;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:XXXX@147.135.12.128;user=phone>;tag=xz13 To: "XXXX"<sip:s@192.168.1.107;user=phone>;tag=as54c1e248 Call-ID: ff01aa-43@147.135.12.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:XXXX@192.168.1.107> Content-Length: 0 to 147.135.12.128:5060 asterisk1*CLI>