[c-nsp] SIP VoIP Config

Wed Apr 9 05:25:05 EDT 2008

Hi Tom

I've managed to get it working, tanks. The working config follow in attach.

Now I've a second issue. The outbound calls are supposed to come from a CT
Server (with a Dialogic D/240SC-T1 card) that connects to the router via a
T1.

During the test phase I'm also using an FXS.

>From the telephone connected to the FXS the call goes fine but from a
telephone connected to the CT server there's a lot of noise added to the
call channel.

Any idea?

Kind regards
Pedro

-----Original Message-----
From: Tom Storey [mailto:tom at snnap.net]
Sent: Tuesday, April 08, 2008 3:39 PM
To: pmatusse at tdm.mz; pmatusse at tdm.mz
Subject: Re: [c-nsp] SIP VoIP Config

The only thing I can see wrong is the following:

001665: *Apr  8 14:41:45.225 PCTime: //-1/xxxxxxxxxxxx/SIP/Msg/
ccsipDisplayMsg:
Sent:
REGISTER sip:Destination_IP:5060 SIP/2.0
Via: SIP/2.0/UDP Source_IP:5060;branch=z9hG4bK5AC47
From: <sip:888....... at Destination_IP>;tag=54447D0-DBD
To: <sip:888....... at Destination_IP>
Date: Tue, 08 Apr 2008 12:41:45 GMT
Call-ID: B9EFB396-48E11DD-A57D8CCE-6E567B30
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1207658505
CSeq: 43 REGISTER
Contact: <sip:888....... at Source_IP:5060>
Expires:  3600
Content-Length: 0

This is your router trying to register with your VoIP provider, but
look at what your VoIP provider is sending back:

001667: *Apr  8 14:41:46.093 PCTime: //-1/xxxxxxxxxxxx/SIP/Msg/
ccsipDisplayMsg:
Received:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP Source_IP:5060;branch=z9hG4bK5AC47
From: <sip:888....... at Destination_IP>;tag=54447D0-DBD
To: <sip:888....... at Destination_IP>;tag=as60705731
Call-ID: B9EFB396-48E11DD-A57D8CCE-6E567B30
CSeq: 43 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Since you do not specify an "authentication" command in your sip-ua
configuration, the router is trying to register the number of your
POTS dial-peer(s). Since the VoIP provider doesnt know about the
numbers you are trying to register (888.......) they are sending back
a 404 to indicate the number is not valid.

You should check with your VoIP provider and see if you have a
username (i.e. phone number) and password you need to specify when
setting up a SIP client, and use an authentication line like I have in
my config.

Tom

On 08/04/2008, at 9:56 PM, <pmatusse at tdm.mz> <pmatusse at tdm.mz> wrote:

> Hi Tom,
>
>
> In attach SIP messages. Note that I've replaced IP Addresses
> with "Source_IP" and "Destination_IP" or "Destination_IP + n on the
> last
> octet".
>
> "Destination_IP + n  on the last octet" means that on the SIP message
> I'm getting de destination SIP gateway address and some oder IPs that
> differ from the destination on the last octet.
>
> Pedro Wiliamo Matusse
> Telecomunicações de Moçambique (TDM)
> DSI
> Tel. +258 21 482820
> Cell. +258 82 3080780
> Fax: +258 21 487812
>
> ----- Original Message -----
> From: <pmatusse at tdm.mz>
> Date: Tuesday, April 8, 2008 1:58 pm
> Subject: Re: [c-nsp] SIP VoIP Config
>
>>
>>
>> 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>
>>>
>>>
>>
> <SIP Messages IP ADD Replaced.TXT>
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