[asterisk-users] local vs remote bridging: no audio

[sean darcy]

Calling into 10.5.0-rc2 from a pstn did provider, I get no audio:

     -- Executing [111 at from-teliax:1] Dial("SIP/teliax-00000010", 
"SIP/office2/+1<number>") in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
     -- Called SIP/office2/+1<number>
     -- SIP/office2-00000011 answered SIP/teliax-00000010
     -- Locally bridging SIP/teliax-00000010 and SIP/office2-00000011

But if I call in over sip from outside with the same number and channel 
all works fine:

     -- Executing [111 at from_11hidden:1] 
Dial("SIP/office_incoming-00000012", "SIP/office2/+1<same_number>") in 
new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
     -- Called SIP/office2/+1<same_number>
     -- SIP/office2-00000013 answered SIP/office_incoming-00000012
     -- Remotely bridging SIP/office_incoming-00000012 and 
SIP/teliax2-00000013

The only difference I can see is Locally vs. Remotely bridging.

sip.conf

nat=yes
directmedia=nonat

Any suggestions appreciated.

sean