Re: Problem with incoming calls on Broadvoiceafter upgrade to 1.4.18

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Problem with incoming calls on Broadvoice after...
\ Jon Miron (16 Mar 2008)
. \ Raj Jain (16 Mar 2008)
. . \ Jon Miron (16 Mar 2008)
. . . \ Raj Jain (16 Mar 2008)
. . . . \ Jon Miron (18 Mar 2008)

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Subject:	Re: Problem with incoming calls on Broadvoiceafter upgrade to 1.4.18
Group:	Asterisk-users
From:	Jon Miron
Date:	18 Mar 2008

Hi Raj,

Sorry for the delay. The NIC in my server running Asterisk died so I
wasn't able to verify until just now. After commenting out the
secret= line, calls go through.

I'll contact their support, but I'm sure they'll be as useless as
ever. This may be the last straw for them.

Thanks again Raj

On Sun, Mar 16, 2008 at 6:44 PM, Raj Jain <rj2807> wrote:
> 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> 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> 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> 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
> > > > 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
> > > > 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
> > > > 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' 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
> > > > 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
> > > > 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
> > > > 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' 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
> > > > 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
> > > > 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
> > > > 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' 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
> > > > 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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> > >
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> > >
> >
> > _______________________________________________
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> >
>
>
>
> --
>
> 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
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