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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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play_and_get_digits - why is minimum 1 digit?
by Nathan Neulinger 20 Sep '13

20 Sep '13

Is there any particular technical reason that the minimum number of digits for play_and_get_digits is 1 instead of 0? There have been a few cases where I was wanting to have "enter a destination phone number followed by pound sign, or enter just pound sign to cancel" type of prompt. Lowering the minimum to zero would accomplish this, but is there some reason not to do that? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Transferring XML metadata as part of SIP INVITE
by Siri MM 20 Sep '13

20 Sep '13

Hi, I have a requirement where, after recording sessions using FS, I need to make a SIP call to a third party device, and then transfer the recorded file to it as part of this call. In order to accomplish this, my plan is to originate a call to this third party SIP device, and bridge the call to my extension, and the dialplan of my extension would playback the recorded file to the third party device. Couple of questions: 1. Any other better ways of achieving this? 2. I also need to transfer the recorded file's metadata in the form of an XML, as part of the SIP INVITE to the device. Although I am aware of adding custom headers using "sip_h", I am not sure how to add a XML file as part of the SIP message. Would appreciate any inputs Thanks!

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