[Asterisk-Users] Jittery meetme conference using Linksys 942 phones

tracinet traci.asterisk at gmail.com
Sat Mar 18 18:21:18 MST 2006


What are your zttest results?  zttest can be run from /usr/src/zaptel/
directory (run ./zttest from there).  Do you have Digium hardware or
ztdummy?

Pedro
http://www.TRACI.net

On 3/18/06, Rana Dutt <astuserlist at gmail.com> wrote:
>
> We have two Linksys 942 phones which sound great when they call each other
> directly through Asterisk. But when they both dial in to a meetme conference
> room, the sound is very jittery. Other phones like Polycom 501 and Snom 360
> sound fine when using meetme.
>
> Both Linksys phones are set to use the default g711u (ulaw) codecs.
> Adjusting the jitter buffer and jitter level settings to various values did
> not help.
>
> We are running Asterisk 1.2.1 on Centos 4.2 (Linux 2.6x kernel) on a
> dual-processor Dell Poweredge 2850 server with 1 Gb RAM. This machine has a
> TE-210 Dual-T1 card plugged in. The meetme.conf file has no general
> settings, just a list of two conference rooms.
>
> Has anyone else experienced sound quality issues with meetme conferences
> using Linksys phones? Any idea what could fix this? Thanks.
>
> Ron
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060318/426e1556/attachment.htm


More information about the asterisk-users mailing list