[Spce-user] Voicemail troubles 2.7 fresh install

[Dave Massey]

Actually just as I posted this I found the issue.
in templates/etc/asterisk/modules.conf.tt2 the codec module codec_gsm.so is not loaded in 2.7, but after some reading, its a dependancy for another module.  I went back to a test 2.6 install, and modules.conf.tt2 has that module loading.  
So I think in the modules.conf.tt2 in the 2.7 templates that module should be loaded as well.  

Cheers!
Dave




On 2013-01-07, at 10:27 AM, Dave Massey <dave at optionsdsl.ca> wrote:


> Ive gone thru all the configs and files, I cant find anywhere where its getting g723 from, my ATAs dont even support it, and this is a fresh install of sipwise, I havent been able to figure this out at all.
> Any one have any ideas?
> 
> Thanks
> Dave
> 
> On 2013-01-03, at 5:47 AM, Andrew Pogrebennyk <apogrebennyk at sipwise.com> wrote:
> 
>> On 01/02/2013 10:20 PM, dave at optionsdsl.ca wrote:
>>> OK I did that, and it fixed that problem,.  But now when someone tries to
>>> leave a message, it will take it, but I get the following log errors, and
>>> the voicemail never gets delivered to the subscriber:
>>> 
>>> Jan  2 15:45:51 sip asterisk[2166]: WARNING[20188]: file.c:200 in
>>> ast_writestream: Unable to translate to format wav49, source format slin
>>> Jan  2 15:45:51 sip asterisk[2166]: WARNING[20188]: translate.c:288 in
>>> ast_translator_build_path: No translator path from unknown to g723
>> 
>> That looks like the user is using g723 and no codec translation is
>> available for the prompts. It should only negotiate alaw or ulaw and the
>> codec transcoding will only be between alaw and ulaw.
>> Please check if you have enabled g723 in your asterisk/sip.conf.
>> By default we have something like
>> disallow=all
>> allow=alaw
>> allow=ulaw
>> If you enable g723 make sure to enable the corresponding transcoding
>> module in asterisk/modules.conf too.
>> 
>> HTH.
> 
> 
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