November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

playAndGetDigits problem in LUA program
by Sanath Prasanna 29 Nov '11

29 Nov '11

Hi all, I used following to get phone number from caller in LUA program. function get_dest() new_number = session:playAndGetDigits(9, 11, 3, 3000, "#", lang_path .. "enter_dest_number.wav", lang_path .. "error.wav", "\\d+"); session:say(new_number, "en", "number", "pronounced"); session:setVariable("dest_numbers",new_number); end -- main -- freeswitch.consoleLog("info", "Voicemail - select_main_menu:\n\r"); reeswitch.consoleLog("info", "Voicemail - Select English :\n\r"); get_dest(); .. .. if caller hang off the phone without entering number(dtmf), I can see lot of same log lines in the freeswitch console log. (around 50000 lines in 1 or 2 sec.Pls see below). *is it bug or just need to ignore. Pls advice. * Br, Sanath 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu: 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - Select English 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - ./vm/en/ 2011-11-22 05:58:23.301593 [INFO] switch_cpp.cpp:1190 Voicemail - select_main_menu:

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removing participant from 3-way bridge
by Royce Mitchell III 29 Nov '11

29 Nov '11

Can I use "uuid_transfer <uuid> park inline" to pull a leg out of a 3-way bridge without breaking the bridge? I want to put that leg back into the callcenter_mod queue once it's done connecting the callers. On Tue, Nov 22, 2011 at 1:22 PM, Anthony Minessale < anthony.minessale(a)gmail.com> wrote: > you can't use break to stop bridging. > you can do "uuid_transfer <uuid> park inline" to put it into an idle state > > > > On Tue, Nov 22, 2011 at 5:54 AM, Nagalenoj H. <nagalenoj(a)gmail.com> wrote: > >> Hi, >> I wanted to cancel the bridge application after issuing it and I tried >> using 'break' application. break application didn't work. But, when I tried >> using playback application followed by break, it worked. >> >> connect >> >> sendmsg >> call-command: execute >> execute-app-name: answer >> >> sendmsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: user/1001 >> >> sendmsg >> call-command: execute >> execute-app-name: break >> >> Logs: >> http://pastebin.freeswitch.org/17843 >> >> Git version as of 18/11/2011. >> >> -- >> Regards, >> Nagalenoj H. >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > 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 > > -- There's a fine line between genius and insanity. I like to use it for dental floss.

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Where is mod_sms?
by Valery Kalinin 29 Nov '11

29 Nov '11

I update Freeswitch from git, but mod_sms not found. modules.conf - not found src/mod/applications/ - not found How to download mod_sms?

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FreeSWITCH Conference Call Tomorrow: Special Guest
by Michael Collins 29 Nov '11

29 Nov '11

Hello all! I just wanted to give everyone a heads up about tomorrow's conference call. We have a special guest scheduled to join us: Jean-Marc Valin from Mozilla! Jean-Marc is a codec guru who is working diligently on the OPUS codec. He will be coming to talk to us about the codec and to answer any questions that you may have. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_30 We look forward to having you on our call tomorrow! -Michael

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Skypopen audio problems
by asdfa asdfasdf 29 Nov '11

29 Nov '11

Hi, i have a little problem i have FS installed, skypeopen_oss client and skypeopen.ko oss sound driver installed. I configure in public a DID number bridge to skypopen/RR/skype22, the call going from DID to SKYPE1 and call SKYPE22, here is the LOG: http://pastebin.freeswitch.org/17877 All work OK, but the sound from DID caller to Skype22 dont reach, no sound, Skype22 to DID caller works ok. Questions: Skype_oss and the skypeopen.ko require a sound card to work? What is the problem? Codecs? Please helpme, thanks and sorry my english.

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Detecting fax on an incoming call leg (direct T.38 case)
by Roland Hänel 29 Nov '11

29 Nov '11

I'm trying to reliably detect whether there is a fax machine on an incoming channel. Right now, it works nicely with the following dial plan: answer spandsp_start_fax_detect sleep 5000 spandsp_stop_fax_detect [... no fax there, so now do stuff that has nothing to do with fax ...] If the remote end is a PSTN fax, spandsp will fire an event upon detecting the CNG tone, and I will react accordingly. However, if the remove end is a T.38 fax, which will immediately re-invite after answer, then the following problem occurs: FreeSWITCH does not answer the re-invite (just says 100 Trying and nothing follows). Even though the remove end fax transmits the CNG tone afterwards, it is not detected by spandsp. I have recorded the call to make sure the CGN tone is there, and it is there. I have even played this tone into another phone call, where it is successfully detected by spandsp. All I can guess is that the non-answered re-invite might block spandsp from detecting the tone? Greetings, Roland

1 1

Anyone got experience with how limit scales with db back end?
by Dave Horton 29 Nov '11

29 Nov '11

I am using limit with db (sqlite) backend to limit all calls into my server. I have 72 inbound channels on gateways sending to my freeswitch, which is running on a single processor dual core 3GHz server, with the sqlite call limits database on the local disk, and at times I am seeing a delay of up to 3 seconds between the arrival of an invite and FS sending a 180 Ringing from the ring ready application. Other delays in signaling I also see intermittently. The only real processing going on before the ring_ready application is executed is the limit application. I see from the code there is a global mutex that looks like will result in all limit checks being single threaded through the code, and I am wondering if I am experience slight intermittent bottlenecks at peak call times. I realize the hash backend is faster and I may switch to that (decision pre-dated my arrival), but I am wondering if anyone out there has any good data about how many cps the limit application with the db backend can scale to. Obviously its dependent on hardware, but I'd still like to get any real world data (or suggestions) that are out there.

2 1

error compiling 1.0.6 on ubuntu 10.10 64 bits
by Valter Nogueira 29 Nov '11

29 Nov '11

after ./configure, make ends with following message gcc -E /home/fastway/freeswitch-1.0.6/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h make: *** [src/include/switch_swigable_cpp.h] Error 1 thanks Valter

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tiff file affects transmission over T.38
by Tim Titov 29 Nov '11

29 Nov '11

First I converted a pdf to tiff with magick and no compression. Then I used tiffcp (3.9.2) to compress it as g3. The file could not be transmitted over T.38 Gafachi or flowroute. It was frustrating to figure this out. I hope others don't run into this problem. P.S.: ghostscript works fine.

2 2

Informal survey: calling multiple endpoints sequentially
by Michael Collins 28 Nov '11

28 Nov '11

Hello all, I'm hoping to get some community feedback as I work on the FreeSWITCH Cookbook. The issue at hand is how you handle the dialing of sequential endpoints. For those of you who do this, would you mind replying to the list and letting us know what you do? I'm curious what methods you use, such as: pipe-sep list of endpoints in a bridge/originate continue_on_fail=true and multiple individual bridge apps mod_lcr or similar Jay Binks' transfer_on_fail method etc. I am particularly interested in the problems that you are trying to solve. In other words - how did the method you chose help you solve your problem? Your perspectives will help me very much. Thanks all! -Michael

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