I don't know if this makes any difference, I created a lot of this configuration with the Asterisk-GUI (SVN-branch-2.0-r4980) and when I edit the users.conf file, there are two entries 'type = peer' for each extension and they are highlighted in red!
Gary Baribault On 06/02/2010 08:32 AM, taimur hasan wrote: > > 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 > <mailto: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. Sign up now. > <https://signup.live.com/signup.aspx?id=60969>
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