June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to disable sending early media ?
by Samira Mh 09 Jun '12

09 Jun '12

<action application="set" data="ignore_early_media=true"/>i have set the above channel variable but when i am going to playback .wav file using lua scripts without issue the seeeion:answer ; i got error :sending early media

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how to play file using lua script within dialplan without need to answer session?
by Samira Mh 09 Jun '12

09 Jun '12

hi guys; i am going to playback .wav file using lua scripts without neet to answer seeeion; how can i do that?

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Re: [Freeswitch-users] mod_directory and LUA for database users
by Perovich Alexander 09 Jun '12

09 Jun '12

Hello, I have been trying to get the mod_directory to query the users that are stored in a database and accessed using a lua script, however every attempt comes back match cannot be found. Any ideas?

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Capture media codes?
by ocset 09 Jun '12

09 Jun '12

Hi Is there a way to capture and display the media codes being sent from a provider. I have been reading up about the various Early Media options but was hoping there was a way to log every tone being received by Freeswitch, to verify what tones a particular provider uses. Thank Dion.

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Freeswitch “real” response to api chat? Having a problem determining result via esl.
by Bill Ryder 09 Jun '12

09 Jun '12

I'm sending an sms message like this over esl (telnet, java client, various methods all yield the same results)(numbers are all dummies): api chat sip|13215555555@6.50.120.201|internal/2395555555@6.50.120.200|test message This works fine with valid numbers. If I put in an invalid number somewhere I may see something like this in the sip trace indicating a problem: U +0.745829 6.50.120.200:5060 -> 6.50.120.201:5060 SIP/2.0 484 Address Incomplete. ... ...however over the esl socket I get the same response event no matter what, just "sent". Also, when I bind to the esl port and attempt to spit out every message possible (no filters), we see most of what comes through a sip trace, but again this bad result sip packet seems not to generate an esl event. Is there a way over esl to probe raw sip messages so that we can test for such a packet manually, or some other way of seeking the final async result? Thanks in advance! Sincerely, William Ryder Network Engineer, Millicorp 12748 University Drive, Fort Myers, FL 33907 Direct: 239-443-5892 Fax: 239-245-9087

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Multilingual IVR in English and Thai
by Abid Saleem 09 Jun '12

09 Jun '12

Hi, Hope to find you well! Does Freeswitch support Multilingual IVR for example English and Thai at the same time. The customer should be given the option to chose either English or Thai. If yes, can you please specify how can it be configured? Thanks in advance

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JS File Operation Error with latest FS
by Jason Moran 08 Jun '12

08 Jun '12

A spidermonkey File operation (often) causes the output of "ASS #####" followed by a "File operation pclose failed" error now that I've done a make current (last update was 4 weeks ago). The code puts a 0 or 1 into the variable fdFoundFile for later testing. This has worked for months until the make current. Now it fails with the pclose error 4/5 times. CODE: console_log("info","-3-\n"); //Do a check to make sure we can read the file var fd = new File("|[ -f "+myFileName+" ] && echo 1 || echo 0"); console_log("info","-4-\n"); fd.open("read"); console_log("info","-5-\n"); var fdFoundFile = fd.read(1); console_log("info","-6-\n"); fd.close(); console_log("info","-7-\n"); LOG: 2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -3- 2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -4- 2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -5- ASS 30937 2012-06-08 15:52:14.717917 [INFO] Testcall.js:1 -6- 2012-06-08 15:52:14.717917 [ERR] Testcall.js:258 Error: File operation pclose failed The # following "ASS" always changes and keeps on increasing. Thanks, Jason M

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Hardware build recommendations
by Timothy Bolton 08 Jun '12

08 Jun '12

Does anyone have any ideas for a good hardware build for a SOHO server? Here's what I'd like running on it: OpenVPN, FreeSWITCH, and a firewall (probably pfSense). My home network doesn't have too much traffic, and I don't get that many calls. -- 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed

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List Admin help
by Sean Devoy 08 Jun '12

08 Jun '12

OK, I see we are stuck with a mailing list and I am very sorry to have brought it up. Fine. Will someone please tell me how to change my "digest" setting? DO NOT TELL ME TO GO TO: http://lists.freeswitch.org/mailman/listinfo/freeswitch-users It does not work. Every effort I make sends me a stupid message saying I am already enrolled. I even assumed I don't know the password. I wrote to: freeswitch-users-owner(a)lists.freeswitch.org that was equally ineffective, but instead of a meaningless reply, I got no reply. Is there some way to get my password or even reset it? Thanks, A very frustrated Sean.

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Audiocodes MP-118 VoIP Gateway can't un-hold
by Mike Kallies 08 Jun '12

08 Jun '12

Hey Everyone, I'm using an Audiocodes MP-118 VoIP Gateway. When making a call through the device to a Freeswitch gateway on the other side, it sends the call to my desk phone just fine. When I mute the call, we get the hold music as per the dial plan, everything is good. Then when I unmute... the hold music stops, and... nothing. The call doesn't drop, it just sits there. In the Audiocodes log, we see: 5d:1h:25m:32s ( lgr_psbrdex)(28879 ) recv <-- acEV_BROKEN_CONNECTION, Ch:0 5d:1h:25m:32s ( lgr_flow)(28880 ) #0:RTP_BROKEN_CONNECTION_EV 5d:1h:25m:32s ( lgr_flow)(28881 ) | #0:RTP_BROKEN_CONNECTION_EV I know this isn't entirely a Freeswitch question, but I figure the crowd here might know the answer or where I should look for one. Calls through our Grandstream device work fine, so I think this can be narrowed down to the Audiocodes device. But maybe there's a different codec or option or something I could try to see if I can make it happy? Thanks, -Mike Kallies

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