Based on the trace alone, it seems like a problem on their end. You may want to try shutting off INVITE authentication (by commenting out secret= line in your sip.conf) to see if the call goes through.
On Sun, Mar 16, 2008 at 6:27 PM, Jon Miron <[EMAIL PROTECTED]> wrote: > Hi Raj, > > Thanks for your response. > > I'm a little confused though. Does this look as if it's a problem > with Broadvoice itself, and not my configuration? Any time I've > called them with problems where it's clearly not my fault (ie nothing > on my end has changed), they're never very helpful. > > > > On Sun, Mar 16, 2008 at 4:45 PM, Raj Jain <[EMAIL PROTECTED]> wrote: > > Looking at the trace, the entity sending you the INVITE is not > > resubmitting INVITE with credentials after the initial INVITE was > > challenged with a 401 response by Asterisk. The trace shows two > > independent calls and both have the same problem. > > > > -- > > Raj Jain > > > > mailto:rj2807 at gmail dot com > > sip:rjain at iptel dot org > > > > > > > > > > On Sun, Mar 16, 2008 at 4:10 PM, Jon Miron <[EMAIL PROTECTED]> wrote: > > > 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 > > > > > > > _______________________________________________ > > -- 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 > > > > _______________________________________________ > -- 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 > -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org _______________________________________________ -- 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
