April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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sounds_dir
by Madovsky 23 Apr '11

23 Apr '11

I have (another stupid) question : where the variable $${sounds_dir} is set ? I see the variable used in vars.xml but not set everywhere in conf directory Thanks

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Re: [Freeswitch-users] Changing From header in voicemail to e-mail notifications
by Joseph L. Casale 23 Apr '11

23 Apr '11

>> 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"/> > >In one of my installs, (1.0.6) that never worked. Is that only valid in trunk? Although the notify-voicemail.tpl & voicemail.tpl utilize ${voicemail_domain} its ignored on my OpenWRT 1.0.6 setup (haven't tested on my CentOS switch. A 'global_getvar' enumerates it as I wanted it... I just manually setup the templates as I needed them... HTH, jlc

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ZRTP Library
by Nasir Iqbal 23 Apr '11

23 Apr '11

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/

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Contents of FreeSWITCH-users digest...
by vibha dear 23 Apr '11

23 Apr '11

thanks Micheal..!! --- On Sat, 23/4/11, freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> wrote: From: freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> Subject: FreeSWITCH-users Digest, Vol 58, Issue 157 To: freeswitch-users(a)lists.freeswitch.org Date: Saturday, 23 April, 2011, 1:53 AM Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: freeswitch xml cdr and csv (Michael Collins) 2. Re: Tuning Up Freeswitch (Antonio Teixeira) 3. Re: freeswitch xml cdr and csv (Ken Rice) _______________________________________________ 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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Need a little bit clarification about gateways
by Boris Kovalenko 23 Apr '11

23 Apr '11

Hello! Am I right, if I created gateway into sofia profile and set register=false I also should permit its IP in the ACL to receive a calls from it? Or there is some param to gateway for automatic acl creation? Like cidr for a user? -- With respect, Boris

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Ringback while NOT YET bridging to leg B
by David Ponzone 23 Apr '11

23 Apr '11

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.

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freeswitch xml cdr and csv
by vibha dear 22 Apr '11

22 Apr '11

hi all..! freeswitch generated cdr in both formats xml an multiline csv format, will there be any parsing overhead in XML parsing while fetching information for reporting and billing? how they will effect the system performance? which module is better to generate CDRs? thanks in advance..!!kasturi :)

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