Dear<div>in normal mode, .call files make a call between the system and who you named remote person, I don't know where are you?</div><div>in natmode=yes, set qualify=yes.</div><div>check the negotiated codecs also.</div>
<div>Best<br><br><div class="gmail_quote">On Sat, Aug 13, 2011 at 1:29 AM, Carlos Chavez <span dir="ltr"><<a href="mailto:cursor@telecomabmex.com">cursor@telecomabmex.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
We are having a problem when trying to use originate or AMI to make a<br>
call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to<br>
call the PSTN. When dialing from IP phones everything works fine. When<br>
you try making the call with originate, AMI or a call file then the<br>
remote person can hear you but you cannot hear them. Why would it<br>
behave differently when dialing from a phone?<br>
<br>
The server is behind NAT and uses externaddr to set the external IP<br>
(static). Anyone had any experience with this?<br>
<br>
Here is my (edited) sip.conf entry:<br>
<br>
[libre-8793]<br>
defaultuser=123456789<br>
secret=XXXXXXXXX<br>
fromuser=123456789<br>
trustrpid=yes<br>
sendrpid=yes<br>
type=peer<br>
fromdomain=<a href="http://i2next.com.mx" target="_blank">i2next.com.mx</a><br>
host=<a href="http://i2next.com.mx" target="_blank">i2next.com.mx</a><br>
nat=yes<br>
qualify=no<br>
insecure=port,invite<br>
directmedia=no<br>
disallow=all<br>
allow=g729<br>
<font color="#888888"><br>
--<br>
Telecomunicaciones Abiertas de México S.A. de C.V.<br>
Carlos Chávez Prats<br>
Director de Tecnología<br>
+52-55-91169161 ext 2001<br>
</font><br>--<br>
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