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
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