Its going to be because you probably need to pre_answer, play a little media before you hair pin the call back out the same provider. :) playback silence_stream://1000 after pre_answer would probably fix it prior to bridge. On Wed, Mar 9, 2016 at 8:32 AM, Peter Steinbach <lists@telefaks.de> wrote:
Hello,
I have the following scenario
- 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 GmbHmailto:lists <lists> (att) telefaks.de Internet: www.telefaks.de
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