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

[Matt Riddell [NZ]]

Brian Roy wrote:
> 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.

The only catalyst to getting it fixed will be if someone posts a bug
entry with full details on bugs.digium.com

If you do, post again here with the ID and discussion and testing can
continue there.

-- 
Cheers,

Matt Riddell
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