• Freeswitch is behind NAT with external profile with ext_sip_ip and ext_rtp_ip
• Echo from an external SIP provider test inbound works with audio
• Calling outbound to the same external SIP provider works with audio
• Receiving a call from the external SIP provider and bridging it to another number at the same external SIP provider does not have audio
• Receiving a call from the external SIP provider and bridging it to a local Patton PRI works with audio

I also set up another (modified NAT) profile

• like external profile - but with a different ext_rtp_ip
• calling outbound to via NAT profile to another SIP provider works with audio
• Receiving a call from the external SIP provider (external profile) and bridging it to another number at the another external SIP provider (nat profile) does not have audio
• Tcpdump btw. does not show an attempt of Freeswitch to issue any outbound RTP packet. I can see only SIP messages towards the outbound SIP provider
  

So I think, there is a problem with briding a natted external call  to another natted external call, both with ext-rtp-ip. It tried this on an older Freeswitch (1 year old) and with a more recent Freeswitch (1 month old), both behave the same. Same scenario with a non-natted Freeswitch works fine.
Btw. Freeswitch logs do show
    RTCPpacket not written
right after bridginbg the call.

Question: Does anybody have a clue, what could cause the problem? Maybe there is any parameter in the profiles, that may help?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de