meetme and Asterisk 1.4.21

[goit]

Hello,
Can someone please let me know if I need to apply meetme patch if I am using Asterisk 1.4.21.1? Users are complaining of bad audio quality whenever a call is connected and number of lines dialed reaches the max (15)

System Details
VERSION: 2.0.5-172
BUILD: 90310-2203
Asterisk: 1.4.21.1
OS: Centos 5.2 (PAE) recompiled with all required settings
CPU: QuadCore 2.0Ghz
RAM: 4GB
number of agents: 10
Trunk: 15 SIP lines dedicated T1 to provider
Load Average: .37
Timing Source: Digium Card (not ztdummy)
Recording: none

Any suggestions on how to address this problem is greatly appreciated.

Thank you

[mflorell]

You should not need to match meetme on 1.4.21.1

What SIP codec are you using?

What kind of bandwidth do you have?

[goit]

Hi Matt,
I am uLaw on my server and phones. There is no transcoding taking place as far as I can tell. I have a dedicated T1 (2MB) to the proivder that is not being used for anything else other than the SIP service. I do not see any issues with bandwidth being completely utilized, with a typical average of less than 850Kbps. The T1 is clean and I've verified proper QoS configurations on the router.

Thanks

[mflorell]

Have you tried another carrier?

How many hops to the carrier?

What is the ping latency to the carrier?

[goit]

I have not tried another carrier yet as I am still trying to figure out if the problem is with my system and how its configured, or it has something to do with the carrier. (process of elemination).

The carrier I am using is local and they specialize in SIP. I have direct access to their SIP servers since I am part of their internal network.

couple of thoughts:

1- Does it make sense for me to downgrade to Asterisk 1.2.30? Do you beleive this might make a difference?

2- I also have the option to switching to a PRI service. Am I going overboard if I do that? I am just worried my problems will continue after this change.

Thank you

[mflorell]

It can't hurt to downgrade to 1.2.30.2, but we also have clients on 1.4.21.2 with no problems going out over SIP so I don't know if that would really fix your issues.

PRIs are great if you have the option, they are guaranteed available and you never have latency issues with them.