Dear in normal mode, .call files make a call between the system and who you named remote person, I don't know where are you? in natmode=yes, set qualify=yes. check the negotiated codecs also. Best
On Sat, Aug 13, 2011 at 1:29 AM, Carlos Chavez <[email protected]>wrote: > 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Pezhman Lali
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
