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

[Gareth Blades]

See 
http://bugs.digium.com/view.php?id=6457

On Tue, 2006-04-18 at 13:11, broadbandvoice at comcast.net wrote:
>  
> I'm running 1.2.3 and that seems to be the most stable version, had
> problems with other versions too.
>         -------------- Original message -------------- 
>         From: "Brian Roy" <mister.roy at gmail.com> 
>         
>         
>         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
>          
>          
>          
> 
> 
> ______________________________________________________________________
> From: Brian Roy <mister.roy at gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud
> Date: Sat, 04 Mar 2006 13:27:53 +0000
> 
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