[Users] Avoid loop on call forwarding

[Jens Carl]

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.

Is there a possibility to avoid these loop? And how realise this?

Best regards
Jens