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

[Brian Roy]

On 3/3/06, Gary Richardson <gary.richardson at gmail.com> 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..
>


I'm running 1.2.1 and most of mine get cut short too. I posted this on the
list a few months ago and nobody had any suggestions. BJ said I should
probably post a bug on it but I haven't had time to continue to troubleshoot
it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been
watching change logs and hadn't seen anything surrounding mixmonitor so I've
let it go.

Please continue to update us if anyone gets some resolution. I'm glad to
know there are lots of us experiencing this. That should be the catalyst to
get it fixed.

-Brian
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