- FreeSWITCH-users - FreeSWITCH Mailing Lists

Not sure of the correct terminology
by ocset 25 Nov '11

25 Nov '11

Hi I am not sure what I should call this topic but I am trying to find out about using Freeswitch in an environment where the caller is prompted with a list of options like "if you know the callers extension, enter it now" etc. Is this possible? What is the correct term for it and where would I find more info on it? Thanks

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Re: [Freeswitch-users] High load on database server
by Raimund Sacherer 25 Nov '11

25 Nov '11

Hello, I have had lot's of issues with 2.14 odbc as well, the Threadong parameter helped to reduce the crashes from approx. 4 a day to 1 a day, but the crashing was still there. I compiled my own debian packages for odbc with the 2.11 centos 5 sources (so I can install other debiacn packages that depend on odbc) and this works like a charm! So I would recommend using 2.11 instead of 2.14, I do not have expierence with newer odbc's though. using postgres 9.(0|1), 8.4 and debian squeeze with fairly recent Freeswitch boxes best ----- Original Message ----- From: "Cliff Wells" <cliff(a)develix.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Thursday, October 27, 2011 8:02:46 AM Subject: Re: [Freeswitch-users] High load on database server On Wed, 2011-10-26 at 11:12 -0500, Anthony Minessale wrote: > if you have older unixodbc edit /etc/odbcinst.ini and add the param > Threading=0 to each db entry. > In general postgres to me appears to care more about multithreaded > environments and with the above threading=0 change works pretty well. I tried this (after googling a bit and coming across an older thread you responded to) and it didn't change anything. Crashes at ~330 concurrent calls like clockwork. This was with unixODBC 2.14 and PostgreSQL 8.4 odbc driver. Regards, Cliff FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 Síguenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para móvil Logitravel.com Este correo electrónico y, en su caso, cualquier fichero anexo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electrónico por error, se ruega notifíquese inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formarán parte de un fichero registrado como CLIENTES con número de inscripción 2070610043 en la Agencia General de Protección de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposición, rectificación y cancelación a estos datos que deberá ejercer mediante escrito a la dirección anteriormente citada.

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Re: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ?
by chrisbware＠interfree.it 24 Nov '11

24 Nov '11

Your INVITE is 1672 bytes. Some router split packet over 1500 bytes. To test if this is the problema, disable all codecs leaving only "basic codecs" as Aastra call them (g711a,g711u,g729) and disable ice. Packet should be smaller: if the problem is fixed in this way, packet fragmentation is the cause. -----Messaggio originale----- Da: Charlie Orford <charlie.orford(a)attackplan.net> Inviato il: 24 Nov 2011 - 13:47 A: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Hi Chris, I've increased the debugging and taken a new trace: http://pastebin.freeswitch.org/17864 And here is the output of tport_sip.log: http://pastebin.freeswitch.org/17865 A person on the #freeswitch IRC channel suggested the problem might be due to packet fragmentation? Charlie On 24/11/2011 14:29, chrisbware(a)interfree.it wrote: > I suspect problem is on SDP but we can't see it on pastebin. > -----Messaggio originale----- > Da: Charlie Orford > Inviato il: 24 Nov 2011 - 10:17 > A: FreeSWITCH Users Help > > > Hi list > > I have done some googling but cannot find much info about what causes FS > to send a SIP/2.0 400 Bad Request response. > > My phones are able to register but when I try a test call (e.g. to 9196) > FS responds to the INVITE with Bad Request. For a short transcript of > this in action see: http://pastebin.freeswitch.org/17863 > > My setup: > I have a clean git head install (cloned yesterday) and am running the > unmodified "out of the box" configuration (except I have enabled > NDLB-force-rport on the internal profile and set a default poassword). > FS is on a public server (no nat) and phones (aastra 57i models) are all > natted behind an ADSL router. > > Thanks & Regards, > Charlie > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------------------------------------------------------- > Valore legale alle tue mail > InterfreePEC - la tua Posta Elettronica Certificata > http://pec.interfree.it > ------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------- Valore legale alle tue mail InterfreePEC - la tua Posta Elettronica Certificata http://pec.interfree.it -------------------------------------------------------------------------------

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Re: [Freeswitch-users] Reasons for a SIP/2.0 400 Bad Request ?
by chrisbware＠interfree.it 24 Nov '11

24 Nov '11

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Re: [Freeswitch-users] wimax double nat
by mazilo 24 Nov '11

24 Nov '11

new_voip wrote > ----------------------------------------------------------------------------------------------------- > *case 1* (does not work) > FS(public static IP) <-- internet <--(public IP ) Nat <-- 10.x.x.x (CPE) > NAT <-- ATA (192.168.x.x) > ----------------------------------------------------------------------------------------------------- Is there a reason for a double NAT/Firewall? If no, then remove the additional layer of NAT/Firewall. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/wimax-double-nat-tp7024457p70… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Playback race condition
by Hynek Cihlar 24 Nov '11

24 Nov '11

Hello all, I got on shaky grounds issuing commands to Freeswitch over ESL. For example, issuing the following commands close enough (on my system at around 100 ms and less) causes troubles. *bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(150,4000,425);loops=-1* *<100ms appart* *bgapi uuid_break 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 all* *previous playback is not stopped, the following is queued (not played)* *bgapi uuid_broadcast 71f8f250-8ad6-4ab3-855a-3cbc71075fe2 playback::tone_stream://%(1000,4000,425);loops=-1* It looks like the first tone playback is not properly initialized when the break arrives. Waiting for the start of the first tone playback (waiting for the right event) is not a solution, I don't want the tone to be played at all if the events come this close. Any ideas? Hynek

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Reasons for a SIP/2.0 400 Bad Request ?
by Charlie Orford 24 Nov '11

24 Nov '11

Hi list I have done some googling but cannot find much info about what causes FS to send a SIP/2.0 400 Bad Request response. My phones are able to register but when I try a test call (e.g. to 9196) FS responds to the INVITE with Bad Request. For a short transcript of this in action see: http://pastebin.freeswitch.org/17863 My setup: I have a clean git head install (cloned yesterday) and am running the unmodified "out of the box" configuration (except I have enabled NDLB-force-rport on the internal profile and set a default poassword). FS is on a public server (no nat) and phones (aastra 57i models) are all natted behind an ADSL router. Thanks & Regards, Charlie

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using break application after bridge
by Nagalenoj H. 24 Nov '11

24 Nov '11

Hi, I wanted to cancel the bridge application after issuing it and I tried using 'break' application. break application didn't work. But, when I tried using playback application followed by break, it worked. connect sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: bridge execute-app-arg: user/1001 sendmsg call-command: execute execute-app-name: break Logs: http://pastebin.freeswitch.org/17843 Git version as of 18/11/2011. -- Regards, Nagalenoj H.

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new git - no more 'SIP auth challenge' messages in log
by Vitalie Colosov 23 Nov '11

23 Nov '11

Hi List, After upgrading to the latest git, I don't see anymore "SIP auth challenge" messages in the log for normal register attempts. Is there any new switch to enable these message? Log verbosity is INFO and <param name="log-auth-failures" value="true"/> Thanks, Vitalie

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FreeSWITCH Conference Call Today
by Michael Collins 23 Nov '11

23 Nov '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_23 There have been a number of changes/additions to mod_conference, so we are going to talk about those today. Also, there are several FreeSWITCH updates to talk about. If we have time and if enough subject matter experts are available we will also talk about SIP presence. See you soon! -Michael

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