On 9/29/05, Iqbal <iqbal@gigo.co.uk> wrote:

whats is sip debug on asterisk showing

Bogdan-Andrei Iancu wrote:

> Hi Matt,
>
> I redirected this email on the users mailing list - it's more
> appropriate.
>
> the idea seams ok, with couple of comments:
>    1) be sure that fwd to localhost is ok (instead of a routable IP)
>    2) doing Record-Route may be a good think.
>
> to debug tour problem, add some log("...") statements into your script
> to be able to trace the processing. Also a network trace (including on
> lo device) will be helpful to see what happens - if the messages are
> received, if they are sent and where. Also watch the log for potential
> errors.
>
> regards,
> bogdan
>
>
>
> Matt L. Zhu wrote:
>
>> has anyone successfully setup openser as the frontend proxy for
>> asterisk? here is my setup
>>
>> /etc/asterisk/sip.conf
>> [general]
>> context=default
>> port=5065
>> bindaddr=0.0.0.0
>> srvlookup=yes
>>
>> [ser]
>> type=user
>> context=proxy
>> host=192.168.0.10
>>
>> then i edited openser.cfg to do something like this
>>
>>            if
>> (uri=~"sip:[a-zA-Z\.]*@(xxx\.xxx\.com)|(192\.168\.0\.10)") {
>>                    forward( localhost, 5065 );
>>                    break;
>>            };
>>
>> i connected two sipphones (wengo) in this case to openser, but calls
>> are not going through at all, connecting directly to asterisk works.
>> have anyone worked in this situation?
>>
>> thanks
>>
>>
>>
>> _______________________________________________
>> Devel mailing list
>> Devel@openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/devel
>>
>
>
> _______________________________________________
> Users mailing list
> Users@openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
> .
>

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