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