<div dir="ltr">Hi jiri,<div><br></div><div>sorry for my late reply.</div><div><br></div><div>If I have understood properly, from your analysis the problem looks related to call routing if the calls comes from a specific network interface.</div><div>I then suggest to investigate in the following direction:</div><div> * capture the sip trace of a working and not working call using sngrep or tcpdump. Then you can compare and check differences. It could be that for some unknown reason the call is routed differently.</div><div> * capture and compare kamailio proxy logs for a working and not working call and try to see if you notice any relevant difference between them.</div><div><br></div><div>Cheers</div><div>Marco</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, May 2, 2024 at 10:01 AM Jiri Ptacnik <<a href="mailto:ptakjura@gmail.com">ptakjura@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">I have confirmed, that problem is network related.<br>
<br>
- When call is going through neth0, then everything works.<br>
- When call is going through ens19, then redirect to voicebox after<br>
specified time is not working (immediate redirect works).<br>
<br>
JP<br>
<br>
<br>
út 2. 4. 2024 v 11:11 odesílatel Jiri Ptacnik <<a href="mailto:ptakjura@gmail.com" target="_blank">ptakjura@gmail.com</a>> napsal:<br>
<br>
><br>
> Voicemail works<br>
><br>
> - when immediate redirect to voicebox is set<br>
> - when ringing some time condition is set AND (!) call is made between<br>
> subscribers on the same SPCE platform<br>
><br>
> When call is originated for example in GSM network and is going do<br>
> SPCE from outside, immediate redirect to voicebox works, conditional<br>
> is NOT working.<br>
><br>
> It does not seems to sounds problem related.<br>
><br>
> How about this networks settings, does it looks right for you:<br>
><br>
><br>
> ---<br>
> hosts:<br>
> self:<br>
> dbnode: '1'<br>
> interfaces:<br>
> - lo<br>
> - neth0<br>
> - ens19<br>
> lo:<br>
> cluster_sets:<br>
> - default<br>
> eee: no<br>
> ip: 127.0.0.1<br>
> netmask: 255.255.255.0<br>
> shared_ip: []<br>
> shared_v6ip: []<br>
> type:<br>
> - sip_int<br>
> - ha_int<br>
> - aux_ext<br>
> - ssh_ext<br>
> - api_int<br>
> - rtp_int<br>
> - stor_int<br>
> v6ip: ::1<br>
> v6netmask: 128<br>
> neth0:<br>
> dns_nameservers:<br>
> - 8.8.8.8<br>
> eee: no<br>
> gateway: 111.111.111.158<br>
> hwaddr: 33:33:33:33:33:33<br>
> ip: 111.111.111.154<br>
> netmask: 255.255.255.224<br>
> type:<br>
> - sip_ext<br>
> - rtp_ext<br>
> - web_int<br>
> - ssh_ext<br>
> - web_ext<br>
> ens19:<br>
> eee: no<br>
> hwaddr: 22:22:22:22:22:22<br>
> ip: 222.222.222.150<br>
> netmask: 255.255.255.252<br>
> post_up:<br>
> - route add -net 333.333.333.48/28 gw 222.222.222.149<br>
> type:<br>
> - sip_ext_incoming<br>
> - rtp_ext<br>
> nodename: spce<br>
> role:<br>
> - proxy<br>
> - lb<br>
> - mgmt<br>
> - rtp<br>
> - db<br>
> - storage<br>
> status: online<br>
> swraiddevices: []<br>
> sysdescr: ''<br>
> syslocation: ''<br>
> sysname: ''<br>
><br>
><br>
> thanks<br>
> JiriP<br>
><br>
><br>
><br>
> po 25. 3. 2024 v 12:16 odesílatel Jiri Ptacnik <<a href="mailto:ptakjura@gmail.com" target="_blank">ptakjura@gmail.com</a>> napsal:<br>
> ><br>
> > When I set language for Voicebox on Domain level, then Asterisk<br>
> > reporting EN for example in Digits sounds, but with same error - no<br>
> > audio available.<br>
> ><br>
> > Also custom greetings upload works in Old admin GUI, not in New GUI.<br>
> ><br>
> > JiriP<br>
> ><br>
> > po 25. 3. 2024 v 9:54 odesílatel Jiri Ptacnik <<a href="mailto:ptakjura@gmail.com" target="_blank">ptakjura@gmail.com</a>> napsal:<br>
> > ><br>
> > > Thanks Marco,<br>
> > ><br>
> > > unfortunately no success.<br>
> > ><br>
> > > - I rised Asterisk debug level, but there is no information about<br>
> > > specific missing file: I can see some messages DE language is in use,<br>
> > > but this is not set for Subsriber, nor in default level (EN is set).<br>
> > ><br>
> > > - I also reset Soundset<br>
> > ><br>
> > > - Also I can see, If i try to setup custom voicemail greetings, these<br>
> > > are not saved or applied, but no error message anywhere. But for<br>
> > > example I have one subscriber from earlier with custom greeting and he<br>
> > > still has his greetings. Also there are probably problems with saving<br>
> > > language settings...<br>
> > ><br>
> > > Overall, this seems not directly Voicemail problem, more deeper<br>
> > > problem "something goes wrong after upgrade from 9.5 do 10.5". In<br>
> > > another question I have problems with redis/keydb, may be, it is some<br>
> > > way related... I am looking into logs, but without luck to see<br>
> > > anything clear...<br>
> > ><br>
> > > service.log: "No pushd defined or found..." after ngcp-status command<br>
> > > sems.log: "Could not extract expires value..."<br>
> > > panel.log: "Failed to find path API/attach", "Error 500..."<br>
> > > redis.log: is always empty, seems strange to me...<br>
> > > syslog: nothing special<br>
> > ><br>
> > > I think, best thing to do could be re-install system. Is there any<br>
> > > best-practice for this? For example install clean SPCE of same version<br>
> > > as now, then copy MySQL DB + NGCP config files to this new system...?<br>
> > ><br>
> > > JiriP<br>
> > ><br>
> > > čt 21. 3. 2024 v 11:38 odesílatel Marco Capetta via Spce-user<br>
> > > <<a href="mailto:spce-user@lists.sipwise.com" target="_blank">spce-user@lists.sipwise.com</a>> napsal:<br>
> > > ><br>
> > > > Hi Jiri,<br>
> > > ><br>
> > > > the announcement played by Asterisk in case of not answer or unconditional forward to voicemail are different.<br>
> > > > Possible tests:<br>
> > > > * Check that your asterisk sound packages are updated<br>
> > > > * 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.<br>
> > > > * Remove all the voicemail custom greetings<br>
> > > > * If you enter in the astersik cli, you can also increase the verbose/debug level and get additional information about the missing file.<br>
> > > ><br>
> > > > Thank you<br>
> > > ><br>
> > > > Regards<br>
> > > > Marco<br>
> > > ><br>
> > > ><br>
> > > ><br>
> > > ><br>
> > > > On Thu, Mar 21, 2024 at 9:22 AM Jiri Ptacnik <<a href="mailto:ptakjura@gmail.com" target="_blank">ptakjura@gmail.com</a>> wrote:<br>
> > > >><br>
> > > >> Hi,<br>
> > > >><br>
> > > >> SPCE 10.5.6<br>
> > > >><br>
> > > >> When Redirect to Voicebox is set up (without condition, immediately),<br>
> > > >> it works fine.<br>
> > > >><br>
> > > >> When the condition is set (for example After 15 seconds redirect to<br>
> > > >> Voicebox), then after 15 seconds ringing, the call is terminated<br>
> > > >> without going into Voicebox.<br>
> > > >><br>
> > > >> In voicemail.log is this:<br>
> > > >><br>
> > > >> Mar 20 19:29:40 spce asterisk[611]: WARNING[4157][C-00000002]:<br>
> > > >> app.c:1941 in __ast_play_and_record: No audio available on<br>
> > > >> SIP/sip_proxy-00000001??<br>
> > > >> Mar 20 19:29:40 spce asterisk[611]: [Mar 20 19:29:40]<br>
> > > >> #033[1;31mWARNING#033[0m[4157][C-00000002]:<br>
> > > >> #033[1;37mapp.c#033[0m:#033[1;37m1941#033[0m<br>
> > > >> #033[1;37m__ast_play_and_record#033[0m: No audio available on<br>
> > > >> SIP/sip_proxy-00000001??<br>
> > > >><br>
> > > >> Any hints? It could be a problem with network interfaces and its Types<br>
> > > >> defined in network.yml?<br>
> > > >><br>
> > > >> thanks<br>
> > > >> JiriP<br>
> > > >><br>
> > > >> --<br>
> > > >> Spce-user mailing list<br>
> > > >> <a href="mailto:Spce-user@lists.sipwise.com" target="_blank">Spce-user@lists.sipwise.com</a><br>
> > > >> <a href="http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com" rel="noreferrer" target="_blank">http://lists.sipwise.com/mailman/listinfo/spce-user_lists.sipwise.com</a><br>
> > > ><br>
> > > ><br>
> > > ><br>
> > > > --<br>
> > > ><br>
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</blockquote></div><br clear="all"><div><br></div><span class="gmail_signature_prefix">-- </span><br><div dir="ltr" class="gmail_signature"><div dir="ltr"><div><span style="font-size:9px"><br></span></div><span style="font-size:9px">Sipwise GmbH | an Alcatel-Lucent Company<br>Leonard-Bernstein-Straße 10 | 1220 Wien<br><a rel="noopener" href="mailto:gfuchs@sipwise.com" target="_blank">Email: invoice@sipwise.com</a><br>Website: <a href="https://www.sipwise.com" target="_blank">https://www.sipwise.com</a><br><br>Particulars according Austrian Companies Code paragraph 14<br>Geschäftsführer: DI Dr. Gernot Fuchs Sitz: Wien FG: Landesgericht Wien FN: 305595f<br>Die
übermittelten Informationen sind ausschließlich dem Empfänger
vorbehalten und können vertrauliches oder privilegiertes Material
enthalten. Ein Gebrauch, Weiterleitung oder Verbreitung in jeglicher
Form seitens Dritter ist nicht gestattet. Sollten Sie diese Nachricht
irrtümlich erhalten haben, bitten wir Sie, sich mit dem Absender in
Verbindung zu setzen und die Nachricht einschließlich der Anlagen zu
löschen. Jegliche nicht autorisierte Nutzung dieser E-Mail ist illegal.<br>Hinweis
zum Datenschutz: Sipwise GmbH ist der Schutz Ihrer Daten wichtig. Auch
durch die Kommunikation per E-Mail werden personenbezogene Daten
ausgetauscht. Erfahren Sie mehr unter: <a rel="noopener" href="http://www.sipwise.com/" target="_blank">www.sipwise.com</a><br><br>Managing Director: DI Dr. Gernot Fuchs HQ: Vienna FG: Vienna Regional Court FN: 305595f<br>Confidentially
notice. This e-mail transmission may contain legally privileged and/or
confidential information. Please do not read it if you are not the
intended recipient(s). Any use, reproduction or disclosure by any other
person is prohibited. If you have received this e-mail in error, please
notify the sender and destroy the original transmission and its
attachments without reading or saving it in any manner. Any unauthorized
use of this e-mail is illegal.<br>Note on data protection: Sipwise GmbH
attaches great importance to the protection of your data. Personal data
is also exchanged through communication by e-mail. Learn more at: <a rel="noopener" href="http://www.sipwise.com/" target="_blank">www.sipwise.com</a></span><br></div></div>