[Asterisk-Users] SIP - Problem with audio clipping

[Michael Welter]

Using a SIP connection with a CLEC, the downstream (received) audio is 
perfect when the mute button is activated on the phone.  However, when 
there is upstream audio (i.e., talking or even breathing into the 
microphone), the downstream audio is cut off.  It's kinda like having a 
half-duplex audio connection.

When I divert outgoing calls to another provider, these calls are fine. 
  However, the inbound calls from the CLEC continue to have clipped audio.

This happens with Polycom, Cisco, and Grandstream phones, so I don't 
believe it's a phone problem.

I have this problem with two different Asterisk system using the same 
CLEC.  One is * v1.2.4 and the other is v1.2.5.  The systems have 
totally different motherboards.

Has anyone had a similar problem, and what was the cause?

Thanks,

-- 
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike at TelecomMatters.net
www.TelecomMatters.net