[Kamailio-Users] OpenSER + Asterisk (Voicemail) Issue....

[Daniel-Constantin Mierla]

Hello,

On 07/30/08 06:58, Gerard A. Matthew wrote:
> I'm having this little issue when implementing the voicemail feature.
>
> My openser.cfg looks like this in the failure route:
>
>
>     if(!t_was_cancelled())
>        {
>          revert_uri();
>          rewritehostport("voicemail.mydomain.com:5061");
>          append_branch();
>
>          #PREVENT SOME CRAZY VOICEMAIL LOOP
>          xlog("L_INFO", "INFO: CALL TO VOICEMAIL");
>          setflag(10);
>          route(1);
>        }
>
>
> On my asterisk end after the time out, i'm viewing the following:
>
>   SELECT * FROM sipusers WHERE name = 'XXXXXX'
>
> i.e  XXXXXX = the PSTN number i'm using to call into the IP phone 
> that's connected to OpenSER
>
> There seems to be a simple mixup with the number that is sent to 
> asterisk. Obviously there is no user with the PSTN number,
> however there is one with the called number.
>
> Any idea as to what wold be causing this? Have I provided enough 
> information?
>

revert_uri() restores the initial R-URI (the one the request came in 
with). If you need to send some other id to voicemail, you can save it 
in an AVP once you discover it in the routing block and use that before 
forwarding to the voicemail system.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla
http://www.asipto.com