June 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Non Blocking Teletone (Background Audio)
by Klaus Teller 26 Jun '08

26 Jun '08

Hi Folks, I was was wondering if it's possible to play a teletone in a non-blocking way. What I want to achieve is following: 1) Party A calls to my FS box, 2) A Javascript script picks up the call 3) The script generates a teletone to party A. 4) Then the script starts some processing while A is on hold listening for the teletone (or any other audio?) 5) The script dials some other party B 6) Party B picks up 7) Script stops the teletone being played to A 8) Script bridges A and B. As far as I could experiment with teletone.generate, the execution of the script is paused when the teletone is played to the near end (Party A). I would love if you could help me with this. Thanks, Klaus L'Imbecile. -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer

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config / scripting questions from voice mail integration attempt
by John Wehle 26 Jun '08

26 Jun '08

The AT&T System 25 sends inband DTMF at the beginning of a call being transferred to voicemail which identifies the extension that was called so I'm creating a javascript to parse the DTMF and do the right thing. Questions: 1) Following the example I found on thelostpacket.org I've created a script and placed it in /usr/local/freeswitch/scripts. I've added: <extension name="system25_vmail"> <condition field="destination_number" expression="^5590$"> <action application="javascript" data="$${basedir}/scripts/system25vmail.js"/> </condition> </extension> to dialplan/default.xml. When I call 5590 freeswitch can't find the script. If I replace $${basedir} with /usr/local/freeswitch, then the script is called. What's the proper way to specify the path to a script in the dialplan? Is there a way to say that it's in the script directory under the freeswitch installation point without using a full path? 2) The DTMF looks something like 123#456# . If I use getDigits(3, '#') followed by another getDigits(3, '#'), then the first call returns 123, however the second call doesn't return 456. It appears that getDigits returns when it either gets the requested number of digits * or * it encounters the terminator. Is that correct? So should I simply do something like getDigits(9, '#') followed by another getDigits(9, '#')? 3) One thought I had was to do something like: <extension name="system25_vmail"> <condition field="destination_number" expression="^5590$"> <action application="javascript" data="$${basedir}/scripts/system25vmail.js"/>  <condition field="system25_vmail_mode" expression="00"> <action application="voicemail" data="check default $${domain} ${system25_vmail_caller}"/> </condition>  <condition field="system25_vmail_mode" expression="02"> <action application="voicemail" data="default $${domain} ${system25_vmail_callee}"/> <action application="send_dtmf" data="#90${system25_vmail_callee}"/> </condition> the idea being that when 5590 is dialed the script will be called to parse the DTMF and set the system25_vmail_XXX variables which are then used by the later conditions. However the wiki for Dialplan XML says: Those accustomed to Asterisk may expect the call to follow the dialplan by executing the applications as it parses them allowing data obtained from one action to influence the next action. This is not the case ... so it sounds like I can't use an action to set variables to be used for the next condition / action. However the example on thelostpacket.org does: <action application="javascript" data="$${basedir}/scripts/read.js conf-getconfno.wav 5"/> <action application="conference" data="${res}@default"/> which appears to do exactly what wiki says you can't so I'm confused. 4) My other thought was to have the javascript call the voicemail application after parsing the DTMF. I.e.: switch (mode) { // Direct Inside Access case "00": if (isNaN (parseInt (from))) { console_log ("err", "Invalid VMAIL calling PDC from PBX\n"); session.flushDigits (); exit(); } session.setVariable ("voicemail_authorized", "false"); apiExecute ("voicemail", "check default " + domain + " " + from); however the apiExecute doesn't seem to invoke voicemail. Probably because there's a difference between invoking an application and an api. So, how do you invoke an application from a javascript called from a dialplan? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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SIP / MixMeeting
by Nick Temple 26 Jun '08

26 Jun '08

Hello, I'm attempting to get Mix Meeting work with freeswitch .... they use IP authorization only, however username / password are required in the config file. If I set them to an arbitray value, Mix returns "forbidden". If I leave them out, then Freeswitch complains that the gateway is invalid. Is there something I may have missed that addresses this? Thanks, Nick

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g.729
by David Villasmil 26 Jun '08

26 Jun '08

Hello all, Is there a licensed g.729 yet? When will we be able to transcode from another codec to g.729?? I know we can pass-through, but transcode? cheers d

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Recent VAD enhancement?
by UV 25 Jun '08

25 Jun '08

I’m trying to send this and it doesn’t seem to post… here’s another attempt: After upgrading from latest stable version for windows (RC4) to latest version, I’ve noticed a significant loss of audio during a call. It almost seems like there’s an enhanced VAD that triggers the audio on and off – although the VAD feature is off. Why VAD? Because as you inject audio on the receiving side, you can then hear incoming audio from the transmitting side. Although this is not noticeable when using tone generation or shout, it’s mostly evident when playing MoH (Music on hold, not Medal of Honor). It sounds like about 90% of the audio is gone. We used identical configuration files for both tests. We’re still investigating this behavior and soon we’ll dive to network level and debug outputs. Anyone else encountered that as well? No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52

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Ast/FS Rosetta Stone
by Michael Collins 25 Jun '08

25 Jun '08

FYI, I create a new wiki page: http://wiki.freeswitch.org/wiki/Rosetta_stone I thought maybe those who know about Asterisk and FreeSWITCH both could start dropping stuff in there. I think that people coming from Asterisk might have an easier time if they had some sort of frame of reference. Note that I put a disclaimer at the top - there isn't really a one-to-one Ast-to-FS translation for most of the stuff, but if you said to an experienced Asterisk user that "conf/dialplan/default.xml" is analogous to "/etc/asterisk/extensions.conf" then he/she would at least have a starting point. I'm hoping to add more stuff like the fact that there aren't really "priorities" in FS but that you can use scripting langs, ivr menus, etc. to mimic a lot of that kind of functionality. Anyway, your thoughts and suggestions are welcomed. I will do my best to add stuff as I am able... -MC

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Meeting time today. 5PM Central
by Brian West 25 Jun '08

25 Jun '08

Don't forget the wiki meeting is at 5pm today. Brian West sip:brian@freeswitch.org

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FS and TDM
by Anton 25 Jun '08

25 Jun '08

Hello! It's interesting to have any info on how good would be FS in the TDM switching, including the proper timing and timing propagation among systems? Also many TDM hardware cards allows in-card frame bridging (so no load on CPU than on the switch) - what with support of this? Or FreeSwitch not so supposed yet to be a core of a TDM switch? Regard, Anton.

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mod_shout.c:497 write_stream_thread() MP3 encode error -3!
by e schmidbauer 24 Jun '08

24 Jun '08

i am trying to stream a conference call via icecast2 i have an extension set up as follows: <action application="record_session" data="shout://source:password@localhost /lecture.mp3"/> <action application="conference" data="fs"/> when i dial the extension, i am able to connect to the conference and the lecture.mp3 stream appears for about 5 seconds and then it disappears. also i get an error message in the freeswitch console as follows: mod_shout.c:497 write_stream_thread() MP3 encode error -3! i've tried many variations of this extension with no success. i am able to play mp3s during the conference and also record the conference to an mp3 so i know that mod_shout is working. any help would be much appreciated. thanks.

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Update to dialplan/default.xml - x9992 = show_info extension
by Michael Collins 24 Jun '08

24 Jun '08

FYI, Brian recently committed a patch that adds a new extension to the dialplan: 9992, aka "show_info". It simply does an info app then hangs up. We added it so that you have a quick and easy means of doing simple testing. For example, if you have a SIP phone registered you can just dial 9992 and see a bunch of information dumped to the CLI. Or you can transfer a call to 9992 and quickly get an info dump that might help you debug a problem. If you haven't modified conf/dialplan/default.xml then you can update to the latest trunk, delete default.xml, then do a make samples and the newest default.xml dialplan file will be ready for use. If you are on IRC then be sure to let noobs know about 9992 because it will save you (and them) lots of hassle when debugging. -MC

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