[asterisk-users] MixMonitor and ChanSpy

[Carlos Alvarez]

It's a good thing I never read that warning, since I've been using those in
a call center environment for about seven years and never had that issue.
 Started with 1.2, went to 1.4 and 1.6 now.  So I can't answer your
question about when it was "fixed" but I've never had a problem doing it
(70 concurrent calls max, all recorded, 5 concurrent channels spied max).



On Tue, Feb 7, 2012 at 5:48 AM, Tiago Geada <tiago.geada at gmail.com> wrote:

> that means that from 1.4.18 that issue is no longer present ?
>
> On 7 February 2012 12:44, Jonas Kellens <jonas.kellens at telenet.be> wrote:
>
>> **
>> On 02/07/2012 01:07 PM, Sammy Govind wrote:
>>
>> Hello,
>>
>>  I've been managing multiple call centres, almost all of them having
>> their calls recorded one way or other. Even in PBX environments with
>> MixMonitor and call recordings I haven't came across the situation where I
>> discovered that I can't chanspy a call because its recorded !
>> Which version of asterisk you are using ! can you paste the CLI logs
>> which show a complete call with a failed attempt to Chanspy ?
>>
>>
>> Using Asterisk 1.6.2.22.
>>
>> The fact that ChanSpy can not be used with MixMonitor is something I read
>> on the wiki :
>>
>> Attention
>>
>>    - Up to and including Asterisk 1.4.17 ChanSpy can cause a *
>>    crash/segfault* if used together with Monitor<http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor>or
>>    MixMonitor <http://www.voip-info.org/wiki/view/MixMonitor> at the
>>    same time. 1.4.18 is supposed to attack this issue by using "audiohooks"
>>    that replaces the current ChanSpy approach.
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Carlos Alvarez
TelEvolve
602-889-3003
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120207/aa5d9e99/attachment-0001.htm>