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

[Pezhman Lali]

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 México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
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-- 
Pezhman Lali
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