[asterisk-users] One way audio when using originate...

[Carlos Chavez]

	We are having a problem when trying to use originate or AMI to make a
call.  We have an Asterisk 1.8.5.0 server which uses a SIP provider to
call the PSTN.  When dialing from IP phones everything works fine.  When
you try making the call with originate, AMI or a call file then the
remote person can hear you but you cannot hear them.  Why would it
behave differently when dialing from a phone?

	The server is behind NAT and uses externaddr to set the external IP
(static).  Anyone had any experience with this?

Here is my (edited) sip.conf entry:

[libre-8793]
defaultuser=123456789
secret=XXXXXXXXX
fromuser=123456789
trustrpid=yes
sendrpid=yes
type=peer
fromdomain=i2next.com.mx
host=i2next.com.mx
nat=yes
qualify=no
insecure=port,invite
directmedia=no
disallow=all
allow=g729

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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