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
