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SIP VoIP Config

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pmatusse at tdm

Apr 7, 2008, 2:08 PM

Post #1 of 7 (2192 views)
Permalink

SIP VoIP Config

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

Attachments:

config HJ3825 07 04 2008 23 00h.TXT (4.45 KB)

ben at internode

Apr 7, 2008, 6:57 PM

Post #2 of 7 (2106 views)
Permalink

Re: SIP VoIP Config [In reply to]

If you haven't already, try posting this in the cisco-voip mailing
list, they are very active, cisco-voip [at] puck

Ben

On 08/04/2008, at 6:38 AM, <pmatusse [at] tdm> <pmatusse [at] tdm> 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
> https://puck.nether.net/mailman/listinfo/cisco-nsp
> archive at http://puck.nether.net/pipermail/cisco-nsp/

_______________________________________________
cisco-nsp mailing list cisco-nsp [at] puck
https://puck.nether.net/mailman/listinfo/cisco-nsp
archive at http://puck.nether.net/pipermail/cisco-nsp/

pmatusse at tdm

Apr 8, 2008, 1:39 AM

Post #3 of 7 (2098 views)
Permalink

Re: SIP VoIP Config [In reply to]

Hi Ben,

Done it already. Thanks

Pedro Matusse

-----Original Message-----
From: Ben Steele [mailto:ben [at] internode]
Sent: Tuesday, April 08, 2008 3:58 AM
To: <pmatusse [at] tdm>
Cc: cisco-nsp [at] puck
Subject: Re: [c-nsp] SIP VoIP Config

If you haven't already, try posting this in the cisco-voip mailing
list, they are very active, cisco-voip [at] puck

Ben

On 08/04/2008, at 6:38 AM, <pmatusse [at] tdm> <pmatusse [at] tdm> 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
> https://puck.nether.net/mailman/listinfo/cisco-nsp
> archive at http://puck.nether.net/pipermail/cisco-nsp/

_______________________________________________
cisco-nsp mailing list cisco-nsp [at] puck
https://puck.nether.net/mailman/listinfo/cisco-nsp
archive at http://puck.nether.net/pipermail/cisco-nsp/

pmatusse at tdm

Apr 8, 2008, 3:51 AM

Post #4 of 7 (2095 views)
Permalink

Re: SIP VoIP Config [In reply to]

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>
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>
<pmatusse [at] tdm> 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
> > https://puck.nether.net/mailman/listinfo/cisco-nsp
> > archive at http://puck.nether.net/pipermail/cisco-nsp/
>
>

_______________________________________________
cisco-nsp mailing list cisco-nsp [at] puck
https://puck.nether.net/mailman/listinfo/cisco-nsp
archive at http://puck.nether.net/pipermail/cisco-nsp/

pmatusse at tdm

Apr 8, 2008, 4:16 AM

Post #5 of 7 (2100 views)
Permalink

Re: SIP VoIP Config [In reply to]

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>
Date: Tuesday, April 8, 2008 1:14 pm
Subject: Re: [c-nsp] SIP VoIP Config

> 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>
> 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>
> <pmatusse [at] tdm> 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>
> > > 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>
> > > <pmatusse [at] tdm> 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
> > >>> https://puck.nether.net/mailman/listinfo/cisco-nsp
> > >>> archive at http://puck.nether.net/pipermail/cisco-nsp/
> > >>
> > >>
> > >
> >
> >
>
_______________________________________________
cisco-nsp mailing list cisco-nsp [at] puck
https://puck.nether.net/mailman/listinfo/cisco-nsp
archive at http://puck.nether.net/pipermail/cisco-nsp/

pmatusse at tdm

Apr 8, 2008, 4:58 AM

Post #6 of 7 (2094 views)
Permalink

Re: SIP VoIP Config [In reply to]

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>
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>
<pmatusse [at] tdm> 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>
> > 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>
> > <pmatusse [at] tdm> 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>
> >>> 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>
> >>> <pmatusse [at] tdm> 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
> >>>>> 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>
>
>

Attachments:

SIP Config Update.TXT (4.88 KB)

pmatusse at tdm

Apr 9, 2008, 2:25 AM

Post #7 of 7 (2095 views)
Permalink

Re: SIP VoIP Config [In reply to]

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]
Sent: Tuesday, April 08, 2008 3:39 PM
To: pmatusse [at] tdm; pmatusse [at] tdm
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.......@Destination_IP>;tag=54447D0-DBD
To: <sip:888.......@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.......@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.......@Destination_IP>;tag=54447D0-DBD
To: <sip:888.......@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> <pmatusse [at] tdm> 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>
> 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>
>> 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>
>> <pmatusse [at] tdm> 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>
>>>> 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>
>>>> <pmatusse [at] tdm> 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>
>>>>>> 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
>>>>>>> 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>
>>>>>> <pmatusse [at] tdm> 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
>>>>>>>> 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>

Attachments:

config 2HJ3825 09 04 2008 10 20h.txt (4.54 KB)

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