[asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18
Raj Jain
rj2807 at gmail.com
Sun Mar 16 17:44:43 CDT 2008
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 <mironj at gmail.com> 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 <rj2807 at gmail.com> 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 <mironj at gmail.com> 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: 320190-32 at 147.135.0.128
> > > 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 - 320190-32 at 147.135.0.128
> > > 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: 320190-32 at 147.135.0.128
> > > 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 '320190-32 at 147.135.0.128' 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: 320190-32 at 147.135.0.128
> > > 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: 7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com
> > > 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: 7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com
> > > 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
> > > '7c356bca35c9e62f6c90c3287466e70a at sip.broadvoice.com' 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: 660240-66 at 147.135.0.128
> > > 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 - 660240-66 at 147.135.0.128
> > > 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: 660240-66 at 147.135.0.128
> > > 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 '660240-66 at 147.135.0.128' 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: 660240-66 at 147.135.0.128
> > > 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!
> > >
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> >
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>
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--
Raj Jain
mailto:rj2807 at gmail dot com
sip:rjain at iptel dot org
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