Ok it seems that the pbx can see that I am recieving a call (or at least my broadvoice number sees it I'm not sure which)
Here is the results of me making a call to my pbx with "sip debug peer bv" (broadvoice) Can someone please take a look at this output, it looks like the call is recieved but either not acted upon or something. All calls get a fast buys and broadvoice claims it isn't them. I have all firewall ports open that need to be (5060-5070 udp+tcp, 10000-20000 udp, 69 udp) The call originates from me (Tim Porritt) to the number registered with Broadvoice (Kira Duckett), any idea of the issue? Everything looks fine as far as I can see. Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: "Porritt Tim"<sip:[EMAIL PROTECTED];user=phone>;tag=9bdf To: "Kira Duckett"<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 33532 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 peer 'bv' 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 147.135.12.250:33532 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) Looking for 2068660133 in from-pstn Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 147.135.12.128:5060 From: "Porritt Tim"<sip:[EMAIL PROTECTED];user=phone>;tag=9bdf To: "Kira Duckett"<sip:[EMAIL PROTECTED];user=phone>;tag=as6fafa40c 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> Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: "Porritt Tim"<sip:[EMAIL PROTECTED];user=phone>;tag=9bdf To: "Kira Duckett"<sip:[EMAIL PROTECTED];user=phone>;tag=as6fafa40c Via: SIP/2.0/UDP 147.135.12.128:5060;received=24.17.77.152 Content-Length: 0 7 headers, 0 lines Destroying call '[EMAIL PROTECTED]' == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 192.168.8.151:5060;branch=z9hG4bK4e5a6712 From: <sip:[EMAIL PROTECTED]>;tag=as3111bfd4 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 107 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:[EMAIL PROTECTED]> Event: registration Content-Length: 0 _______________________________________________ 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
