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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