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

Mon Mar 25 07:16:11 EDT 2024

When I set language for Voicebox on Domain level, then Asterisk
reporting EN for example in Digits sounds, but with same error - no
audio available.

Also custom greetings upload works in Old admin GUI, not in New GUI.

JiriP

po 25. 3. 2024 v 9:54 odesílatel Jiri Ptacnik <ptakjura at gmail.com> napsal:
>
> 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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