[Users] Avoid loop on call forwarding

[Klaus Darilion]

Jens Carl wrote:
> Hay list,
> 
> I've got following scenario: I have call forwarding with the help of 
> OpenSER. There are no problems with forwarding within the SIP network. 
> Also the forwarding to a PSTN destination is possible if the caller is 
> from the SIP network. And also the forwarding to a SIP destination if 
> the call comes from a PSTN destination.
> 
> The only problem is when the caller is from the PSTN network and the 
> callee tries to forward this call to another PSTN destination.
> 
>        one server     one server
>        Asterisk*      OpenSER
>           |              |
> call: 12  | call: SIP43  |
> --------->|------------->| look for forwarding and find 56
>           |              | makes new branch with new found R-URI
> call: 56  |     call: 56 | relay the call to the PSTN gateway
> <---------|<-------------|
>           |              |
> 
> * ASTERISK works as gateway (incoming and outgoing calls to PSTN)
> 
> That should be the the chain of the call but the OpenSER/Asterisk 
> detects a loop and the call is dropped. But this will be the most used 
> option of your call forwarding functionality.

This is a known limitation of Asterisk. Asterisk detects this "spiraled" 
call falsely as "looped" call.
> 
> Is there a possibility to avoid these loop? And how realise this?

You can try the uac module to change the From header. Maybe this way you 
can fool asterisk's loop detection algorithm.

Another possibility is to setup the call forwarding in Asterisk (or both 
in Asterisk and ser).

regards
klaus

> 
> Best regards
> Jens
> 
> _______________________________________________
> Users mailing list
> Users at openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users