[Spce-user] [EXTERNAL] Voicemail and No audio available on SIP

[Jiri Ptacnik]

Thanks Marco,

unfortunately no success.

- I rised Asterisk debug level, but there is no information about
specific missing file: I can see some messages DE language is in use,
but this is not set for Subsriber, nor in default level (EN is set).

- I also reset Soundset

- Also I can see, If i try to setup custom voicemail greetings, these
are not saved or applied, but no error message anywhere. But for
example I have one subscriber from earlier with custom greeting and he
still has his greetings. Also there are probably problems with saving
language settings...

Overall, this seems not directly Voicemail problem, more deeper
problem "something goes wrong after upgrade from 9.5 do 10.5". In
another question I have problems with redis/keydb, may be, it is some
way related... I am looking into logs, but without luck to see
anything clear...

service.log: "No pushd defined or found..." after ngcp-status command
sems.log: "Could not extract expires value..."
panel.log: "Failed to find path API/attach", "Error 500..."
redis.log: is always empty, seems strange to me...
syslog: nothing special

I think, best thing to do could be re-install system. Is there any
best-practice for this? For example install clean SPCE of same version
as now, then copy MySQL DB + NGCP config files to this new system...?

JiriP

čt 21. 3. 2024 v 11:38 odesílatel Marco Capetta via Spce-user
<spce-user at lists.sipwise.com> napsal:
>
> Hi Jiri,
>
> the announcement played by Asterisk in case of not answer or unconditional forward to voicemail are different.
> Possible tests:
>  * Check that your asterisk sound packages are updated
>  * Try to switch to another of the available voicemail language and see if ti solves the problem. In this case you are most probably missing one of the announcements.
>  * Remove all the voicemail custom greetings
>  * If you enter in the astersik cli, you can also increase the verbose/debug level and get additional information about the missing file.
>
> Thank you
>
> Regards
> Marco
>
>
>
>
> On Thu, Mar 21, 2024 at 9:22 AM Jiri Ptacnik <ptakjura at gmail.com> wrote:
>>
>> Hi,
>>
>> SPCE 10.5.6
>>
>> When Redirect to Voicebox is set up (without condition, immediately),
>> it works fine.
>>
>> When the condition is set (for example After 15 seconds redirect to
>> Voicebox), then after 15 seconds ringing, the call is terminated
>> without going into Voicebox.
>>
>> In voicemail.log is this:
>>
>> Mar 20 19:29:40 spce asterisk[611]: WARNING[4157][C-00000002]:
>> app.c:1941 in __ast_play_and_record: No audio available on
>> SIP/sip_proxy-00000001??
>> Mar 20 19:29:40 spce asterisk[611]: [Mar 20 19:29:40]
>> #033[1;31mWARNING#033[0m[4157][C-00000002]:
>> #033[1;37mapp.c#033[0m:#033[1;37m1941#033[0m
>> #033[1;37m__ast_play_and_record#033[0m: No audio available on
>> SIP/sip_proxy-00000001??
>>
>> Any hints? It could be a problem with network interfaces and its Types
>> defined in network.yml?
>>
>> thanks
>> JiriP
>>
>> --
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>
>
>
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