Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell).
When I enable debugging I get the following: SIP Debugging Enabled for IP: 147.135.0.128 net-xero*CLI> <--- SIP read from UDP://147.135.0.128:5060 ---> INVITE sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: "Toronto ON"<sip:<my cell #>@147.135.0.128;user=phone>;tag=prtu To: "<my name>"<sip:s@<servers IP>> Via: SIP/2.0/UDP 147.135.0.128:5060 Contact: <sip:<my cell #>@147.135.0.128:5060> Supported: 100rel Content-Length: 309 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.250 t=0 0 m=audio 28274 RTP/AVP 0 8 18 96 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:97 t38/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (10 headers 14 lines) --- == Using SIP RTP CoS mark 5 Sending to 147.135.0.128 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user '<my cell #>' in SIP users list Found peer 'sip.broadvoice.com' for '<my cell #>' from 147.135.0.128:5060 net-xero*CLI> <--- Reliably Transmitting (no NAT) to 147.135.0.128:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128 From: "Toronto ON"<sip:<my cell #>@147.135.0.128;user=phone>;tag=prtu To: "<my name>"<sip:s@<servers IP>>;tag=as77a74c13 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r106946 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06b61489" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) net-xero*CLI> <--- SIP read from UDP://147.135.0.128:5060 ---> ACK sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: "Toronto ON"<sip:<my cell #>@147.135.0.128;user=phone>;tag=prtu To: "<my name>"<sip:s@<servers IP>>;tag=as77a74c13 Via: SIP/2.0/UDP 147.135.0.128:5060 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:9020 sip_reregister: -- Re-registration for <my Broadvoice #>@sip.broadvoice.com REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 147.135.0.128:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP <servers IP>:5060;branch=z9hG4bK339bbe4e;rport Max-Forwards: 70 From: <sip:<my Broadvoice #>@sip.broadvoice.com>;tag=as2a457f50 To: <sip:<my Broadvoice #>@sip.broadvoice.com> Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX SVN-trunk-r106946 Expires: 120 Contact: <sip:s@<servers IP>> Event: registration Content-Length: 0 --- net-xero*CLI> <--- SIP read from UDP://147.135.0.128:5060 ---> SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER From: <sip:<my Broadvoice #>@sip.broadvoice.com>;tag=as2a457f50 To: <sip:<my Broadvoice #>@sip.broadvoice.com> Via: SIP/2.0/UDP <servers IP>:5060;branch=z9hG4bK339bbe4e Contact: <sip:s@<servers IP>> Expires: 30 Event: registration Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:14949 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) net-xero*CLI> <--- SIP read from UDP://147.135.0.128:5060 ---> INVITE sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: "Toronto ON"<sip:<my cell #>@147.135.0.128;user=phone>;tag=ikmn To: "<my name>"<sip:s@<servers IP>> Via: SIP/2.0/UDP 147.135.0.128:5060 Contact: <sip:<my cell #>@147.135.0.128:5060> Supported: 100rel Content-Length: 309 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.250 t=0 0 m=audio 28276 RTP/AVP 0 8 18 96 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:97 t38/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (10 headers 14 lines) --- == Using SIP RTP CoS mark 5 Sending to 147.135.0.128 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user '<my cell #>' in SIP users list Found peer 'sip.broadvoice.com' for '<my cell #>' from 147.135.0.128:5060 net-xero*CLI> <--- Reliably Transmitting (no NAT) to 147.135.0.128:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128 From: "Toronto ON"<sip:<my cell #>@147.135.0.128;user=phone>;tag=ikmn To: "<my name>"<sip:s@<servers IP>>;tag=as6ef11459 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r106946 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a011874" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) net-xero*CLI> <--- SIP read from UDP://147.135.0.128:5060 ---> ACK sip:<my Broadvoice #>@<servers IP>:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: "Toronto ON"<sip:<my cell #>@147.135.0.128;user=phone>;tag=ikmn To: "<my name>"<sip:s@<servers IP>>;tag=as6ef11459 Via: SIP/2.0/UDP 147.135.0.128:5060 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- sip.conf: register => <username>:<password>@sip.broadvoice.com [sip.broadvoice.com] type=peer user=<username> host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=<username> secret=<password> username=<username> insecure=very context=from-bv authname=<username> dtmfmode=inband dtmf=inband canreinvite=yes extensions.conf: [from-bv] exten => s,1,Answer() exten => s,n,MusicOnHold exten => <number>,Answer() exten => <number>,n,MusicOnHold I did these 2 lines for debugging purposes. the dialplan is a little more complex but because this didn't even work, there's no point in posting. Does anyone have any idea why this works fine when I was using 1.2 but suddenly with 1.4.18 it isn't? This is on a server connected directly to the internet, no NAT. Nothing else has changed on it, and Link2Voip (SIP) and Vittelity (IAX) works flawlessly. Any help would be GREATLY appreciated. Thanks in advance! _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
