September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Checking call is connected while bridged in lua [SOLVED]
by David Villasmil 23 Sep '13

23 Sep '13

Hello Steven, That's a great suggestion!! Works like a charm :) Thanks! David On Tue, Sep 24, 2013 at 12:04 AM, Steven Ayre <steveayre(a)gmail.com> wrote: > Combine lua with execute_on_answer perhaps. > > Or create an outbound session and only bridge it to Aleg when answered > (though then you'll need to generate your own ringback) > > > > On Monday, September 23, 2013, David Villasmil wrote: > >> Hello Brian, >> >> I'm receiving a call on one side and bridging it to the B side. But when >> the calls is answered on the B-side I need to actually check the B-side is >> the intended terminating party, so I´m adding a sip header and the >> receiving party is storing the header in a db that I check... >> >> I'm not sure if i'm clear enough... >> >> thanks! >> >> David >> >> >> On Mon, Sep 23, 2013 at 2:59 PM, Brian West <brian(a)freeswitch.org> wrote: >> >>> What exactly are you trying to accomplish? >>> >>> On Sep 22, 2013, at 8:18 PM, David Villasmil < >>> david.villasmil.work(a)gmail.com> wrote: >>> >>> > Thanks for replying Brian, >>> > >>> > But as far as I remember, the execution of the script would only >>> continue "after" the "bridge" finished (with answer, busy, whatever)... if >>> so, I can't check the status on the bridge WHILE the bridge is ongoing... >>> is this right? >>> > >>> > Thanks >>> >>> >>> >>> -- >>> Brian West >>> brian(a)freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH_Wire , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> > _________________________________________________________________________ > 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 > >

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Freeswitch RTP endpoint linked to another RTP endpoint
by Danny Gershman 23 Sep '13

23 Sep '13

So I have an INVITE that comes in that doesn’t have SDP in it. I have the “enable-3pcc” turned on so FreeSWITCH does respond properly. So then it responds with SDP allocating an endpoint. Then I take that SDP information and I turn around and send a new INVITE with the SDP in it. The hopes is that this endpoint is then bridges to a new endpoint. Essentially Freeswitch is connected to itself. Here is the pastebin - http://pastebin.freeswitch.org/21327 Look at the CreatePlayConnectionForConference:1 in the User-Agent field. And then look at the CreatePlayConnectionForConference:2. Those two are supposed be linked together with RTP. Danny Gershman Principal Engineer Cinchcast 475 Park Avenue South, 19th Floor New York, NY 10016 646.807.0840 (o) 973.931.6239 (c)

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Lua version 5.1 to version 5.2
by Antonio Silva 23 Sep '13

23 Sep '13

Hi, I was trying to use bitwise functions inside a lua script and just realize that is not supported under fs... Because the new bitwise operations are only available for the new version 5.2 in the new table bit32. Locking into the code, the version used in FS the lua release 5.1.4. Is it going to update to the present version 5.2? Or is there a way for me to compile it this new lua version? i can tested Thanks,

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Webapi url formation issue
by Mark Chaney 23 Sep '13

23 Sep '13

I am trying to setup a webapi url that I can form to make automated calls to an extension and then do some text to speech. I cant seem to get the url right and could really use some help. Here is what I have so far: http://pbx.domain.com:8080/webapi/originate?sofia/internal/644@pbx.domain.c… But I get a response of: -ERR NO_USER_RESPONSE The extension never rings though and I dont see anything in the console to show that my custom dialplan was ran: <extension name="text2speech" > <condition field="destination_number" expression="^text2speech-(.*)$" > <action application="answer" /> <action application="sleep" data="2000" /> <action application="playback" data="shout://translate.google.com/translate_tts?tl=en&q=$1" /> <action application="sleep" data="10000" /> <action application="hangup" /> </condition> </extension> Any suggestions? Thanks for your help. I really appreciate it. This is my first attempt at using the webapi, so I am a bit lost. -Mark

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On-conference distortion for FreeTDM channels, but not SIP channels
by Bob Hartwig 23 Sep '13

23 Sep '13

I've tried everything I can think of on this, so now I'm reaching out to you good people. I have a FreeSWITCH system that has the latest version of FreeTDM, Dahdi, libpri, and a Rhino R4T1 quad T1 board with up-to-date firmware. When I call into one of the PRI channels, a "playback"-played file sounds clean, but when I go into conference, the conference audio is distorted. This does not happen with a SIP channel. I would classify the distortion as a "crackle", but it is very regularly periodic. Repeating ticks at (roughly) 10-20 Hz repeat rate, I would say. I can only hear them when audio is present, i.e. the audio seems to be modulated with the ticks. And that conference audio can be the MOH guitar recording. I've eliminated the following: - T1 clocking - Echo Cancellers (I've turned on and off all combinations of hardware and software EC) - T1 board firmware (it's the latest, and the problem only happens on-conference). - Packet length - it's the same (20 mS) for the conference engine, FreeTDM, and Dahdi - mu-law vs. a-law encoding - behaves identically. - Comfort noise - happens with and without. - conference sampling rate (tried 8000 and 16000, even though I'm using TDM, which is 8000). - FreeSWITCH version - happens on 1.2 stable, the latest master branch, and an older version that we were using. If you haven't heard of anything like this, could you please give me your thoughts on why this could happen only on-conference, not while playing WAV files, and only with FreeTDM channels? Thanks!

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Voicemail + Ring time + Anonymous call
by Jerome LEGRAND 23 Sep '13

23 Sep '13

Hello everyone! I walk slowly in the discovery of FreeSWITCH and I can do what I want in general. I have few points on which I block ... - VoiceMail: I normally set my voicemail to my extension using : <param name="vm-enabled" value="true"/> but impossible to operate. If I call, I cannot leave a message ... I do not understands the operation of it :-/ - Ring time: In my inbound configuration I have 60 seconds of ringing, but despite my call timeout, it does not work. I have only 25 seconds. <action> application="set" data="continue_on_fail=true"/> <action> application="set" data="call_timeout=60"/> <action> application="bridge" data="sofia/private/100%${domain_name}"/> <action> application="voicemail" data="default $${domain} 100"/> Probably a configuration error, but I cannot find ... any idea? - Anonymous: I wish I could make outbound anonymous calls, but no idea to do that ... My inbound calls are anonymous (the SIP trace) but my phone still displays the number ... A field problem between caller number and caller name? For now I'm stuck on it ... Do you have any ideas or tips? Thank you in advance! Sincerely, Jerome LEGRAND Bonjour à tous ! J'avance petit à petit dans la découverte de Freeswitch et j'arrive à faire ce que je veux en général. J'ai ces quelques points sur lesquels je bloque ... - VoiceMail : J'ai normalement activé ma messagerie pour mon extension, à l'aide de : <param name="vm-enabled" value="true"/> Mais impossible de la faire fonctionner. Si j'appel, je ne peux pas laisser de message ... Je ne comprend pas bien le fonctionnement de celle-ci :-/ - Ring time : J'aimerais obtenir 60 secondes de sonnerie, mais malgré mon call timeout, cela ne fonctionne pas. <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=60"/> <action application="bridge" data="sofia/private/100%${domain_name}"/> <action application="voicemail" data="default $${domain} 100"/> Surement une erreur de configuration, mais je ne trouve pas ... une idée ? - Anonymous : J'aimerais pouvoir faire des appels en anonymous, mais aucune idée pour faire cela ... Mes appels entrant sont bien anonymous ( sur la trace SIP ) mais mon téléphone affiche quand même le numéro ... Un problème de champ caller number et caller name ? Pour le moment je bloque là-dessus ... Avez-vous des idées ou des conseils ? Merci d'avance ! Cordialement, Jérome LEGRAND

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Issue when dialling number with *
by smrdoshi 23 Sep '13

23 Sep '13

Hi Guys, I am setting up check voicemail feature with *97. When I am dialling *97 from extension I am getting below error, 2013-09-23 18:39:27.621209 [ERR] switch_regex.c:101 COMPILE ERROR: 0 [nothing to repeat][*97] Here is my dialplan, <context name="default"> <extension name="*97"> <condition field="destination_number" expression="*97"> <action application="set" data="effective_destination_number=*97"/> <action application="voicemail" data="check default 192.168.1.13 1234567"/> </condition> </extension> </context> Can body please help me to figure out the issue? Thanks in advance. ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-when-dialling-number-wi… Sent from the freeswitch-users mailing list archive at Nabble.com.

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What does 8000310061 mean?
by Karl Schmidt 23 Sep '13

23 Sep '13

Grepping the source didn't help.. The extension linked to below plays DTMF 8000310061 instead of a ringback. (an 800 number?) I'm trying to get spandsp_start_fax_detect working. I figured out by trial and error that it only works if I first answer the call. That there is no audio for the answered line is the likely problem. I tried pre_answer which fails to detect the fax. How do I play a ringback ? (Needs to stop if an extension answers. ) http:/xtronics.com/uploads/faxextension.txt -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 The first panacea for a mismanaged nation is inflation of the currency; the second is war. Both bring a temporary prosperity; both bring permanent ruin. Ernest Hemingway --------------------------------------------------------------------------------

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mod_managed - beginner's question about an event listener module
by Rafal Gwizdala 21 Sep '13

21 Sep '13

I've looked at mod_managed documentation and can see how to write an API command or a dialplan application, but could you give me a hint how to implement an automatically starting 'background' service that will listen to FS events? The module should start automatically at FS startup and should react to conference-related events. I've looked through Dave Kompel's examples but didn't find such example (or didin't understand the code well enough) Thanks RG

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Issue with auth-calls false
by Trevor Benson 20 Sep '13

20 Sep '13

I am having trouble creating a working unregistered external gateway in FS. I have set auth-calls to false, and have also added an ACL into domains including the cidr of the gateway in question. The after the Invite the FS box always sends a 407 Proxy Auth in return to the gateway. SIP Trace, external, internal & gateway profile are in http://pastebin.freeswitch.org/21443 All suggestions welcome. Thanks, Trevor

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