-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I want to know whether freeSWITCH is able to be a b2bua proxy for both sip and rtp or not. sofia has a registration section to Register freeswitch to one or more upstream VoIP provider. I understand this feature in that way, that UAC registered to freeswitch can use this sip accounts to do outbound calls into a external VoIP-Cloud/domain. My picture of it is as follows: 1. uac1@mydomain.com makes an outbound call to uac2@externaldomain.com 2. freeswitch gets this call, answered it and sets up a RTP stream to uac1. 3. freeswitch uses the destination address of uac1 and establish a call as a member of externaldomain.com eg. freeswitch@externaldomain.com. 4. freeswitch bridges the RTP stream between uac1 and freestwitch's own call. This means uac1 is only talking to freewitch while freeswitch is talking to uac1 destination. So freeswitch acts like a man in the middle. Am I wrong ? In my test environment freeswitch registers to the upstream provider, but when I start a call from uac1 to a uac in that upstream provider domain. freeswitch takes it and send an invite to upstream. In this invite message I can find a source sip address which is that from uac1 instead of freeswitch's registration account for upstream domain. The upstream sip proxy sends first a Trying and then a 488 message "Invalid Session Description". The 488 message contains a Warning named " Warning: 301 xxx.xxx.xxx.xxx 'invalid transport IP address'" I think this reffers to the forwarded uac1 source sip address, so that I think that the upstream sip proxy doesn't accept source addresses which doesn't belog to that upstream sip domain. Any ideas how I can convince freeswitch to act as a UAC-Proxy for rtp and sip? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5UTR4tZeNddg3dwRApXDAJ427t8eg7mr0nG+e7PlSEucrROuQQCfYAzg Wje678MWuowzbN1mE212hHo= =mF5Q -----END PGP SIGNATURE-----