Also check the codecs as if you are using g729 or g723, there is a chance that
they are not available in codecs directory ( /usr/lib/asterisk/modules).
-THQ- !!!ONE
Date: Tue, 1 Jun 2010 19:24:41 -0400
From: zisha...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] no sound between extensions
Do you agree something is blocking the audio in one direction? Can you do a
'rtp debug' and then initiate a SIP call and see if there is two way audio
traffic. Also make sure these extensions have 'canreinvite=no'.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-06-01 7:02 PM, "Gary Baribault" <g...@baribault.net> wrote:
As I stated, the incoming calls are on TDM DS0s connected to the Digium
card, and the extensions are on the same local network as the Asterisk
server. There is currently no NAT anywhere.
Gary Baribault
On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
>
> Output of 'iptables -L -n' would also be helpfu...
--
_____________________________________________________________________
-- 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
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969--
_____________________________________________________________________
-- 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
