[Spce-user] Voicemail troubles 2.7 fresh install
Andrew Pogrebennyk
apogrebennyk at sipwise.com
Thu Jan 3 05:47:56 EST 2013
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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