- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_radius_cdr
by Camila Troncoso 25 Apr '11

25 Apr '11

Hi, I´m working with radius accounting using the mod_radius_cdr module. I´m having trouble with the date format that Radius server receive. An example of what the radius server receive is: “Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"” This date format is very difficult to read and I want to change it to make the accounting easier. I search all around for some param or configuration that allows me to do so, but I only find that this date format is define in mod_radius_cdr.c module. Please some help with this matter. Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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Эхо на Белоруссию
by Boris Kovalenko 25 Apr '11

25 Apr '11

Добрый день! 25.04.2011 20:26:06 +375292718781 - на удаленном конце (в Белоруссии) слышали собственное эхо. -- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005

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Can FreeSWITCH play MoH from remote server?
by Иван Чистяков 25 Apr '11

25 Apr '11

Can FreeSWITCH play MoH from remote server?

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g729 annexa/annexb interoperability
by Javier Gallart 25 Apr '11

25 Apr '11

Hello all we're using a Sangoma D100 card for transcoding. Our configuration, as far as the codec policy is concerned, basically does this: -In the first offer to the b-leg we use the same codec list we receive (<action application="export" data="absolute_codec_string=${ep_codec_string}"/>) -If the call fails with status 488, we repeat the call using all the codecs available. This includes changing from annexb to annexa and viceversa. Let's suppose a-leg only supports g729 annexA, and B-leg only supports G729 annexB. My understanding is that those different codec flavours don't interoperate; in fact I've experienced many times audio problems when trying to set up a call between endpoints supporting different g729 variants. In the described case, FS sends initially annexb=no, and B-leg rejects it with cause 488. We rebuild the offer with annexb=yes, and then the offer is accepted by the B-leg. Our concern is that no transcoding resources are used in this case, and we might run into audio problems because of that. The other concern is that in the answer to the A-leg (in 183 and 200), annexb=yes is included. I'm not sure if all devices would support different fmtp parameters in the offer and the answer (the RFC, as usual, won't be explicit about this). Is there any way to force the transcoding in a situation like the one I described? Thanks in advance Javi

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No Dial Tone ... Zaptel + FreeSwitch + TDM400P
by Pablo Silva 25 Apr '11

25 Apr '11

Hi! I've tryed for severals hours to make a phone call from freeswitch server, using TDM400P. I don't know why i can't hear the dial tone, and it's impossible make a phone call.., i read freeswitch wiki page about TDM400P configuration, as you can see that's my configuration an log http://pastebin.com/Y94UqN5R if I focus on information from the log, you can see that loads openzap, the server recognizes the TDM400P card, the call is routed to the PSTN but no concrete achievements. When you call, I hear the tone, not the dialing, I guess I'm near the solution to my problem, I request your cooperation. In advance, thanks. -Pablo

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Build and install FS with new customized folder on CentOS 5.5
by fieldpeak 25 Apr '11

25 Apr '11

i added a folders(e.g. new_folder) inside the FS source code folder, inside the new_folder there are some files, like below, ./freeswitch/conf ./freeswitch/bin ./freeswitch/*new_folder*/files1, files2... i need this folder and the files under it will be installed when i run 'make install', however, it did not. the commands i run as below, i assume bootstrap.sh will run autoscan which will find my new_folder... ./bootstrap.sh make make install anyone can help how to realize it? thanks. Regards, Charles

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HOLD - signals from phones? How to play MoH?
by Иван Чистяков 25 Apr '11

25 Apr '11

In a bridge mode, when i push a button hold, FreeSWITCH don't playing MoH. But if I set hold through FS console it's works.

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Re: [Freeswitch-users] setup an enviroment for implementating / testing IVR
by Frankie Yiu 24 Apr '11

24 Apr '11

Thanks for the input. But dialing 5000 would be testing the demo IVR, but my question is that how do I make an incoming call (Do I have to dial a extension number?) so that I can hear my own custom audio? Sorry for the confusing/dumb question. Thank you. Frankie > > ---------- Forwarded message ---------- > From: Philippe Le Toquin <philippe(a)ppmt.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 21 Apr 2011 21:27:10 -0400 > Subject: Re: [Freeswitch-users] setup an enviroment for implementating / > testing IVR feature > Hello, > > This link is a good start to get to do things with your new Freeswitch > > http://wiki.freeswitch.org/wiki/Getting_Started_Guide > > to test IVR just dial 5000 from your softphone to access a demo IVR > > One thing to change is the default password of the extension to make it a > bit harder to crack :) > Look in the vars.xml file > > /Philippe > > On 11-04-21 08:45 PM, Frankie Yiu wrote: > > Hi there, > > I would like to start implementating an IVR feature, but the first thing I > need to do is to able to use a softphone to dial into my freeSWITCH so that > an audio would start playing. > Right now I have the original configuration of freeSWITCH which I am not > sure how to dial into the system using the softphone. For example, what > extension I have to call; or if I need to change anything in dialplan, > or configuration, etc. > > I have C# code that let me make outgoing call to my softphone, play audio > or collect DTMF. But I am not sure what to do for incoming call using the > original configuration from freeSWITCH. > > Thanks in advance. > > Frankie > > > _______________________________________________ > FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… > > > > ---------- Forwarded message ---------- > From: Boris Kovalenko <boris(a)tagnet.ru> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 22 Apr 2011 08:01:36 +0600 > Subject: [Freeswitch-users] variable direction > Hello! > > I found that with gateways I may use <variable .... direction="..."/>. > Is this possible with regular (directory) users too? If not, may somebody > explain why? > > -- > С уважением, > Борис Коваленко > ЗАО "Тагнет" > тел. +7 (3435) 230001 > факс +7 (3435) 230005 > > > > > > ---------- Forwarded message ---------- > From: David Ponzone <david.ponzone(a)ipeva.fr> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 22 Apr 2011 04:04:21 +0200 > Subject: [Freeswitch-users] Ringback while NOT YET bridging to leg B > Hi all, > > I am trying to achieve something weird, I admit. > I would need to receive a call from A, then wait X seconds before bridging > to B, but I'd like to send back a ringback to A while sleeping X seconds. > The point is to allow customers to have a nice ringback, which the > beginning of HAS to be heard by callers (like an announce in a call-center). > > A (ugly) way to do that would be to have 2 sound files: one to be played > when answering with playback, and then another one for the ringback. I would > then not use sleep at all, but the bridge would only happen when the first > file is over, achieving quite the same result. > The drawback of this is that I need to ask customers for 2 sound files, and > that can be a pain with most of them. > > I tried all combinations I could think of with ringback, instant_ringback, > with no luck. > > Any ideas ? > > David Ponzone Direction Technique > email: david.ponzone(a)ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pièces jointes sont confidentiels et établis à > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autorisée est interdite. Tout message électronique est susceptible > d'altération. **IPeva** décline toute responsabilité au titre de ce > message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas > destinataire de ce message, merci de le détruire immédiatement et d'avertir > l'expéditeur.* > * > * > > > > > > ---------- Forwarded message ---------- > From: Yehavi Bourvine <yehavi.bourvine(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 22 Apr 2011 07:11:34 +0300 > Subject: Re: [Freeswitch-users] Changing From header in voicemail to e-mail > notifications > Hello John, > > You have to set the following in vars,xml: > > <X-PRE-PROCESS cmd="set" data="voicemail_account=*username*"/> > <X-PRE-PROCESS cmd="set" data="voicemail_domain=*your-domain.com*"/> > > Regards, __Yehavi: > 2011/4/21 John Platts <john_platts(a)hotmail.com> > >> We see FreeSWITCH mod_voicemail in the From header of the voicemail to >> e-mail notifications. We also want to change the E-mail address in the From >> header to come from a specific domain. How do we go about changing the From: >> header in the voicemail to e-mail notifications? We do not want FreeSWITCH >> mod_voicemail to appear in the From header of the e-mail messages. >> >> >> _______________________________________________ >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Nasir Iqbal <nasir(a)ictinnovations.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Fri, 22 Apr 2011 08:44:32 +0500 > Subject: [Freeswitch-users] ZRTP Library > Hi, > > As zfone download server is offline, is there anyone who can provide me > libzrtp SDK source? > > > Nasir Iqbal > > ICT Innovations > http://www.ictinnovations.com/ > > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Thu, 21 Apr 2011 22:49:26 -0700 > Subject: Re: [Freeswitch-users] variable direction > The direction variable is set at the time of the call based upon whether or > not the user is the caller (outbound) or callee (inbound). > > -MC > > On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris(a)tagnet.ru> wrote: > >> Hello! >> >> I found that with gateways I may use <variable .... >> direction="..."/>. Is this possible with regular (directory) users too? >> If not, may somebody explain why? >> >> -- >> С уважением, >> Борис Коваленко >> ЗАО "Тагнет" >> тел. +7 (3435) 230001 >> факс +7 (3435) 230005 >> >> >> _______________________________________________ >> 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 >> > > > _______________________________________________ > 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 > >

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one-way audio problem on some inbound gateways, but not others (and no outbound gateways)
by Fraser Redmond 24 Apr '11

24 Apr '11

I'm getting a one-way-audio problem on 2 of my inbound gateways. I have 3 gateways (1800 numbers), all directing into the same dialplan, which then directs the caller to an extn (answered by a softphone.) One of the gateways works fine, the other two do not (they used to several months ago, I'm not exactly sure when they stopped working.) One of the gateways that doesn't work is used for outbound calls without any problem. The pstn caller's audio arrives at the softphone, but the softphone's audio doesn't make it to the caller. It also doesn't show up in call-recordings. I've tried 3 different softphones to make sure it wasn't a problem with that. (The microphone indicators on them indicate they're picking up audio. Sending a dtmf does go through.) I've just upgraded to the latest version of freeswitch to make sure it wasn't that. Any ideas of what to try? Cheers, Fraser

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OpenVox B100P BRI PTP on FS(libpri/dahdi)
by Christian Benke 23 Apr '11

23 Apr '11

Hi! Has anyone gotten a OpenVox B100P or other cheap HFC-S based ISDN-card to work with freeswitch with BRI point-to-point? I get signalling errors and line-status is DOWN("ERR] ftmod_libpri.c:128 Don't know how to SABME on a type 0 node"). I'm using libpri-1.4.12-beta3 and dahdi 2.3.0.1(The last version supported by the OpenVox patch). I've opened a ticket with OpenVox(see http://bbs.openvox.cn/viewthread.php?tid=1543 for more details) and they've been responsive, but my feeling is that this is not going anywhere soon. Regards Christian

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