[c-nsp] SIP VoIP Config

Tue Apr 8 07:58:12 EDT 2008

Going to send "debug ccsip messages" out put.

"session  
> target sip-server". Is sip-server actually what you have in there, 
> or  
> do you normally have an IP address?

Not sure, I'm in Africa and have SIP gateway in US.

In attach the updated SIP config.

Pedro Wiliamo Matusse
Telecomunicações de Moçambique (TDM)
DSI
Tel. +258 21 482820
Cell. +258 82 3080780
Fax: +258 21 487812

----- Original Message -----
From: Tom Storey <tom at snnap.net>
Date: Tuesday, April 8, 2008 1:35 pm
Subject: Re: [c-nsp] SIP VoIP Config

> Can you turn off all debugging, and then turn on "debug ccsip  
> messages" and forward that to me.
> 
> I also notice that in your dial-peer 100 config you have "session  
> target sip-server". Is sip-server actually what you have in there, 
> or  
> do you normally have an IP address?
> 
> Can you send through a more recent copy of your SIP configuration?
> 
> 
> On 08/04/2008, at 8:44 PM, <pmatusse at tdm.mz> 
<pmatusse at tdm.mz> wrote:
> 
> > Hi Tom,
> >
> > sending again
> >
> >
> > Pedro Wiliamo Matusse
> > Telecomunicações de Moçambique (TDM)
> > DSI
> > Tel. +258 21 482820
> > Cell. +258 82 3080780
> > Fax: +258 21 487812
> >
> > ----- Original Message -----
> > From: Tom Storey <tom at snnap.net>
> > Date: Tuesday, April 8, 2008 1:22 pm
> > Subject: Re: [c-nsp] SIP VoIP Config
> >
> >> I dont see any attached files ?
> >>
> >> On 08/04/2008, at 8:21 PM, <pmatusse at tdm.mz>
> > <pmatusse at tdm.mz> wrote:
> >>
> >>> Hi Tom
> >>>
> >>>
> >>> Thank you. Adapted you config but still no working.
> >>>
> >>> Can you please have a look on the debug output in attach.
> >>>
> >>> Kind Regards
> >>>
> >>> Pedro Wiliamo Matusse
> >>> Telecomunicações de Moçambique (TDM)
> >>> DSI
> >>> Tel. +258 21 482820
> >>> Cell. +258 82 3080780
> >>> Fax: +258 21 487812
> >>>
> >>> ----- Original Message -----
> >>> From: Tom Storey <tom at snnap.net>
> >>> Date: Tuesday, April 8, 2008 10:55 am
> >>> Subject: Re: [c-nsp] SIP VoIP Config
> >>>
> >>>> Hi.
> >>>>
> >>>> If it helps, I recently configured a 1760 to connect to my ISPs
> >>>> VoIP
> >>>> service, and this is the config I used for my sip-ua:
> >>>>
> >>>> sip-ua
> >>>> authentication username 08xxxxxxxx password xxxx
> >>>> no remote-party-id
> >>>> registrar ipv4:1.2.3.4 expires 3600
> >>>> sip-server ipv4:1.2.3.4:5060
> >>>> !
> >>>>
> >>>> Initially I had issues where my calls didnt appear to be dialled
> >>>> via
> >>>> the VoIP provider, but with a bit of debugging from both ends we
> >>>> figured out that I had to "no" the "remote-party-id" feature,
> >>>> hence
> >>>> you see "no remote-party-id" line in my config.
> >>>>
> >>>> The symptoms of my issue were I would dial the number, and it
> >>>> would
> >>>> sit there as if it were waiting for more characters, or it was
> >>>> trying
> >>>> to dial, and would eventually time out. It turns out it was
> >>>> actually
> >>>> dialling the number, but my VoIP provider was rejecting the call.
> >>>>
> >>>> You can use "debug ccsip" to see SIP messages to/from your
> > router,
> >>>>
> >>>> this can help to get clues about what it going on (beware 
> that SIP
> >>>> is
> >>>> quite chatty, so a lot of output can be produced at times).
> >>>>
> >>>> For reference, my dial-peers/voice-ports look like this:
> >>>>
> >>>> voice-port 3/0
> >>>> cptone AU
> >>>> timeouts interdigit 4
> >>>> timeouts call-disconnect 2
> >>>> timeouts wait-release 10
> >>>> description ** FXS right **
> >>>> !
> >>>> dial-peer voice 100 pots
> >>>> destination-pattern 08........
> >>>> port 3/0
> >>>> !
> >>>> dial-peer voice 200 voip
> >>>> destination-pattern [0,1][2-4,7,8]........
> >>>> session protocol sipv2
> >>>> session target ipv4:1.2.3.4
> >>>> dtmf-relay sip-notify rtp-nte
> >>>> signal-type ext-signal
> >>>> codec g711alaw
> >>>> no vad
> >>>> !
> >>>>
> >>>> Other than the config above, I have zero other config related to
> >>>> voice
> >>>> on this router - no translation rules, codec profiles, etc - the
> >>>> above
> >>>> two snips of config are it!
> >>>>
> >>>> My setup is working 100% fine, inbound and outbound.
> >>>>
> >>>> Hope that helps. :-)
> >>>>
> >>>> Tom
> >>>>
> >>>> On 08/04/2008, at 6:38 AM, <pmatusse at tdm.mz>
> >>> <pmatusse at tdm.mz> wrote:
> >>>>
> >>>>> Hi There,
> >>>>>
> >>>>>
> >>>>> Trying to make calls from a POTS do VOIP in SIP setup in 
attach,
> >>>> calls> from POTS are not beeing forwarded to VoIP port.
> >>>>>
> >>>>> Can any one help
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> Pedro Wiliamo Matusse
> >>>>> Telecomunicações de Moçambique (TDM)
> >>>>> DSI
> >>>>> Tel. +258 21 482820
> >>>>> Cell. +258 82 3080780
> >>>>> Fax: +258 21 487812
> >>>>> <config HJ3825 07 04 2008 23
> >>>>> 
00h.TXT>_______________________________________________
> >>>>> cisco-nsp mailing list  cisco-nsp at puck.nether.net
> >>>>> https://puck.nether.net/mailman/listinfo/cisco-nsp
> >>>>> archive at http://puck.nether.net/pipermail/cisco-nsp/
> >>>>
> >>>>
> >>>
> >>
> >>
> > <SIP Call Debug.TXT><SIP Call Debug 2.TXT>
> 
> 
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