[Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud

[Johnathan Corgan]

Gary Richardson wrote:

> I'm running 1.2.4 and just about every call is cut short. I'm using
> Cisco IP phones as end points. All the outbound calls are routed via SIP
> through a PRI line attached to a Cisco 2811..

For me, all incoming/outgoing calls arrive/leave via IAX2/ilbc and all
the local end points are SIP/ulaw (SPA-841s.)

Still haven't seen any recordings cut short on 1.2.4.  By now if I were
using 1.2.1, I would have seen it at least once or twice.

-Johnathan