December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Recommended ULIMIT settings problem
by Bob 17 Dec '11

17 Dec '11

Hi, As per the performance docs, I am trying to set the recommended ULIMIT of 8192 stacksize on a 64 bit machine. I am using the current version of Freeswitch (as pulled off GIT yesterday, Friday). I am running Scientific Linux 6.1 : Linux 2.6.32-131.21.1.el6.x86_64 #1 SMP Tue Nov 22 14:15:09 CST 2011 x86_64 x86_64 x86_64 GNU/Linux My present ulimit settings are as follows : $ ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) unlimited max locked memory (kbytes, -l) unlimited max memory size (kbytes, -m) unlimited open files (-n) 999999 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) unlimited real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited However when I start freeswitch "bin/freeswitch -nc -nonat -nonatmap", I get the following : Error: stacksize 8192 is too large: run ulimit -s 240 from your shell before starting the application. auto-adjusting stack size for optimal performance... Surely this is incorrect, because I am running a 64 bit kernel, 240 is the recommended value for 32 bit ?

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Stable Branch Maintainer Wanted [Employment Opportunity]
by Anthony Minessale 17 Dec '11

17 Dec '11

Hello, We have been fortunate enough to be in a circumstance where we have the proper funding from a generous supporter to advance an energetic and logistically gifted individual from the ranks of our community to become our very own Stable Branch Maintainer. This is a challenging job helping to put together a stable release and other community maintenance tasks. Anyone who is interested, please send references and other relevant info to jobs(a)freeswitch.org Thanks, -- 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

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Coders Wanted Perl/C/JS [Employment Opportunity]
by Anthony Minessale 16 Dec '11

16 Dec '11

Barracuda Networks is hiring developers right now for the CudaTEL product. As you may already know, CudaTEL is the FreeSWITCH-powered PBX appliance designed by the core FreeSWITCH team. We are looking for developers for a few positions with some combo of the following skills: Programmer in the following languages: C Perl Javascript/Jquery Database skills: Postgres We need someone who lives in the United States and preferably someone already in or willing to be located in MI CA WI OK TX Please respond with CV/Resume to jobs(a)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

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FreeTDM does not work
by Valery Kalinin 16 Dec '11

16 Dec '11

I upgraded freeswitch from git. Nothing changed in the configuration! Suddenly stopped working freetdm. Why? my configs: freetdm.conf [span zt pri] name => pri trunk_type => E1 group => e1group b-channel => 1-15 d-channel => 16 b-channel => 17-31 wanpipe.conf [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 freetdm.conf.xml <configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="7"/> </settings> <libpri_spans> <span id="1" name="pri"> <param name="node" value="network"/> <param name="switch" value="euroisdn"/> <param name="l1" value="alaw"/> <param name="dp" value="international"/> <param name="debug" value="all"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans> </configuration> log: 2011-11-27 14:35:12.284837 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 3 2011-11-27 14:35:12.284847 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 4 2011-11-27 14:35:12.284898 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 1 2011-11-27 14:35:12.284906 [ERR] ftmod_wanpipe.c:240 Failed to open wanpipe device span 1 channel 2 2011-11-27 14:35:12.285290 [WARNING] ftmod_zt.c:333 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 1 fd 50 (Invalid argument)] 2011-11-27 14:35:12.285523 [WARNING] ftmod_zt.c:333 this ioctl fails on older ftdmtel but is harmless if you used ztcfg [device /dev/dahdi/channel chan 3 fd 50 (Invalid argument)] 2011-11-27 14:35:12.285634 [ERR] mod_freetdm.c:2465 Error finding FreeTDM span id:1 name:pri Card Digium TE121 Please HELP!

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Lua - origination to local endpoint then bridge to either local or remote destination: no audio
by Alex Crow 16 Dec '11

16 Dec '11

Hi all, I have a Lua script that I am calling from XML RPC from a browser (also same behaviour from the console). The script originates a session to one extension, when that is picked up it originates another session to another endpoint and then bridges the two sessions. The issue is that if the first freeswitch.Session(<endpoint>) has the endpoint local to the FS box, there is no audio at all after the bridge is executed. If the first session is a remote endpoint, it works OK apart from the fact that that endpoint does not hear ringing, but when the second session is answered all is well. If both endpoints are remote, everything is perfect. Here is my Lua script - the no-audio version: local calling_user = argv[1]; local called_num = argv[2]; local session1 = freeswitch.Session("user/"..calling_user); session1:sleep(500); session1:answer(); if (session1:ready() == true) then local session2 = freeswitch.Session("[origination_caller_id_number=02023493482]sofia/gateway/10.10.0.2/" .. called_num); session2:answer(); if (session2:ready() == true) then freeswitch.consoleLog("INFO","Bridging\n"); freeswitch.bridge(session1, session2); end end This one works fine: local calling_user = argv[1]; local called_num = argv[2]; local session1 = freeswitch.Session("[origination_caller_id_name="..called_num.."]sofia/gateway/10.10.0.2/" .. calling_user); session1:answer(); if (session1:ready() == true) then local session2 = freeswitch.Session("[origination_caller_id_number=02076084900]sofia/gateway/10.10.0.2/" .. called_num); --either the above or the below both work in this script, except the below does not present ringing to session1 --local session2 = freeswitch.Session("[ringback=%(400,200,400,450);%(400,2200,400,450)]user/" .. called_num); session2:answer(); if (session2:ready() == true) then freeswitch.consoleLog("INFO","Bridging\n"); freeswitch.bridge(session1, session2); end end Where gateway 10.10.0.2 is a Mitel 3300 connected to the PSTN via ISDN30. I can provide logs if required - please advise as this is driving me bonkers! Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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pagd and dynamic conferences
by Rodney 16 Dec '11

16 Dec '11

I finally got pagd working for a dynamic conference using transfer from IVR but i want to know if there is a way to do this from within another static conference.: I use <control action="execute_application" digits="#" data="execute_extension 759 xml default"/> from within a static conference, it transfers me to the proper conference but the caller looses control and has to hangup to call back to get another option. I want them to have the same caller-controls and be able to hangup back to ivr just like the static conferences work. if i have to use a separate caller control group that is fine, but as it stands this only works from main ivr, it is very important for me to have a direct transfer to a dynamic conference pagd from within a static. does this have something do with clearing the bindings to get new ones? i tried clear digit action but don't think it works for the hard coded caller controls. maybe i am using the wrong method to transfer to the extension 759 from within conference. all help is appreciated. condition destination_number ^759$ 1 action answer 2 action clear_digit_action conf 3 action set conference_user_list=| 11 action play_and_get_digits 4 4 3 5000 # conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 15 action phrase spell,${target_num} 16 action conference ${target_num}-127.0.0.1@default

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freeswitch.email () failing
by Drew Terenzini 16 Dec '11

16 Dec '11

Good evening, I'm attempting to use the freeswitch.email from http://wiki.freeswitch.org/wiki/Lua#freeswitch.email. I am running FS 1.04 on Ubuntu 10.04 LTS, and for various reasons having to do with programs running on top of FS, I cannot just upgrade FS without breaking other things. I have installed postfix successfully and tested sendmail from the Linux CLI without a problem. Emails are delivered to the appropriate external addresses. My switch.conf.xml has: -> <param name="mailer-app" value="sendmail"/> <param name="mailer-app-args" value="-t"/> . and I have a stripped down test script called "email.lua" in my scripts directory as follows: freeswitch.email("<destination(a)destination.com>","<source(a)freeswitch.source. com>","subject: Test message from VTAPENG\n","Hello you have a test message.") The destination and source email addresses are valid and receiving email when sent from other programs/sites. In the FS_CLI, I attempt to test-run the script by entering "lua email.lua", only to get the following error each time: -ERR encounterd freeswitch@vtap> 2011-12-15 16:36:19.246622 [ERR] mod_lua.cpp:182 /opt/freeswitch/scripts/email.lua:1: attempt to call field 'email' (a nil value) stack traceback: /opt/freeswitch/scripts/email.lua:1: in main chunk This error appears to terminate the script, and no response is received from postfix, and no error is logged to mail.log. I've tried a number of ways to modify the outcome, without real result. Searching on the Wiki and the mailing lists hasn't resulted in anything that has worked either. Is freeswitch.email() not supported in FS 1.04? Is there some context that I'm missing for this to work? Any help would be appreciated, thanks!

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freeswitch logic before dialplan
by Miha Zoubek 16 Dec '11

16 Dec '11

Hi, how to disable one variable before it hit dialplan for incoming calls? regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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Question Answertime variable?
by Thomas Hoellriegel 16 Dec '11

16 Dec '11

Hi all, Is there a possibility to determine the caller time for a channel at the end of a call? For example: in asterisk exists a variable: ${ANSWEREDTIME} I like to store the Answertime in a Database for example: Dailly minutes to call a Cellphone. Can your help plese? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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Freeswitch RxFax with T38 & Javascript
by sharad 16 Dec '11

16 Dec '11

Hi All With xml dialplan, Rxfax with T38 is working very well. I am trying to do it differently - Means I want to take the call from xml to Javascript & from Javascript I want to answer the call & switch to T38. Plz let me know if someone has done it Thanks in advance. Sharad

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