Carlos Chavez
2011-Aug-12 20:59 UTC
[asterisk-users] One way audio when using originate...
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 M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110812/9d768815/attachment.pgp>
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 <cursor at telecomabmex.com>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 M?xico S.A. de C.V. > Carlos Ch?vez Prats > Director de Tecnolog?a > +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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110813/1363e473/attachment.htm>