[Spce-user] Voicemail troubles 2.7 fresh install

[Dave Massey]

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.