[Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info

[Martin Joseph]

On Dec 21, 2005, at 2:31 PM, Matt Roth wrote:

> List users,
>
> I have some additional information related to the dropped audio.
Huh, I have noticed this type of popping on an SIP to SIP connection 
using ulaw also,  but I figured it was just me.  I am running * 1.21.

I am kind of a newbie to asterisk, so how should I go about documenting 
this in your opinion.

I don't think mixing down with sox is acceptable as that introduces 
another potential source of noise?

I'd love to help fix this, or pin it down, but would need some more 
direction to do so...

Marty