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TEST
by Brian West 20 May '14

20 May '14

IGNORE -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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OS X Wiki updated for 10.9.3 and new ldns prereq
by Mario G 19 May '14

19 May '14

For those interested: https://confluence.freeswitch.org/display/FREESWITCH/Installation+and+Setup… OS X 10.9.3 tested 10.6 through 10.9 tested with new ldns prerequisites, also tested with updated brew packages The OS X FreeSWITCH installer was updated to include ldns: https://confluence.freeswitch.org/display/FREESWITCH/Installation+on+OS+X+u… Links to git clone 1.2 were corrected (odd since I got them from the Linux page and they worked when tested) now using the same as Linux page git clone for 1.2. Mario G

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ClueCon 2014 Call For Speakers! Hurry Hurry! Time is running short!
by Ken Rice 19 May '14

19 May '14

ClueCon 2014 is shaping up to be the best ClueCon Ever! We are rapidly approaching the dead line for getting those Talk Proposals in! If you have not already submitted a talk proposal, please email it to marketing(a)cluecon.com! We have excellent space this year at The InterContinental Chicago. So be sure to get those proposals in! What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon loves technical presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: * ClueCon talks are 30 minutes in length, including Q&A time with the audience (Lightening Talks are also open! 10 minutes talk, 5 minutes Q&A) * ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio * Attendees enjoy hearing about projects built with open source tools, especially those in a production environment * Highly technical discussions that show the nuts and bolts are especially well-liked * The audience appreciates seeing and participating in live demonstrations We are especially interested in Security-related talks and demonstrations Please send your proposals to marketing(a)cluecon.com. Be sure to include the following items: * Working title * Brief description of the talk (abstract) * Name of the presenter * Bio and Headshot of Presenter * Presenters Contact information (including mobile phone the presenter will have with them at the conference) Don't delay! There are a limited number of openings. We will contact you as soon as your talk has been approved and will inform you of the scheduled time. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Barge in on Shared Line
by Darcy Primrose 19 May '14

19 May '14

Does anyone know if there is a way to disable bargein on shared lines. I use Snom and Grandstream phone, Shared line works perfect on these sets, I would like to disable bargein and give a true key system type of operation where by the third party can only join by invitation. Darcy voice2net

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Uptime/Monitoring/Notification
by Blake Priddy 19 May '14

19 May '14

Do any of you freeswitchers have a recommendation of a service to monitor gateway reg or freeswitch status. Something that can have a web interface and SMS/Email alerts. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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Problems reading DTMFs sent by A
by Helena Garcia-Nieto 19 May '14

19 May '14

Hi All! I am trying to read some DTMFs dialed by A after freeswitch picks the call. My aim is to recollect the DTMFs and create a bridge between the incoming call and the extension he dialed but I am not able to get the dialed DTMFs. I've tried read and play and get digits action with no success. It seems none of the applications I used on the dialplan is able to read the sent dtmfs. I am using it on the public dialplan since the phone is getting to that dialplan form port 5080. I tried also transferring the call to the default dialplan and using the actions there with no success either. <extension name="Read Example"> <condition field="destination_number" expression="^(334)$"> <action application="answer"/> <action application="play_and_get_digits" data="2 5 1 7000 # $${base_dir}/conf/outgoingcall.wav $${base_dir}/conf/outgoingcall.wav myvar \d+"/> <action application="log" data="INFO helena2 -1-${mybar}-"/> </condition> </extension> I've traced on the server and I can see the RTP event packet with the digits getting the server but in my console log I can see there were no info got: 2014-05-12 15:01:27.211903 [NOTICE] mod_dptools.c:1225 Channel [sofia/external/1000@XX.XX.XX.XX:5080] has been answered 2014-05-12 15:02:02.331453 [INFO] mod_dptools.c:1595 helena2 -1-- 2014-05-12 15:02:02.331453 [NOTICE] switch_core_state_machine.c:262 sofia/external/1000@ XX.XX.XX.XX:5080 has executed the last dialplan instruction, hanging up. 2014-05-12 15:02:02.331453 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/external/1000@ XX.XX.XX.XX:5080 [CS_EXECUTE] [NORMAL_CLEARING] I've tried with the read action as well: <extension name="try2" continue="true"> <condition field="destination_number" expression="^(333)$"> <action application="answer"/> <action application="sleep" data="1"/> <action application="read" data="0 10 $${base_dir}/conf/outgoingcall.wav res 10000 #"/> <action application="phrase" data="spell,${res}"/> <action application="log" data="INFO HELENA --${res}__"/> </condition> </extension> 2014-05-12 15:16:51.971820 [NOTICE] sofia_glue.c:4360 Pre-Answer sofia/external/1000@ XX.XX.XX.XX:5080! 2014-05-12 15:16:51.971820 [NOTICE] mod_dptools.c:1225 Channel [sofia/external/1000@ XX.XX.XX.XX:5080] has been answered 2014-05-12 15:17:16.171867 [INFO] mod_dptools.c:1595 HELENA --__ I cannot hear the played wav from my softphone, although I can see in the server trace that it is sent, so maybe I have some NAT problem for the RTP but since I can see the RTP EVENT getting the freeswitch server I don't think it is affecting. I've tried configuring the softphone to send RFC2833 RTP events and also to send the dTMF in band. Both cases the result on the log was an empty variable. I am pretty new at freeswitch so I guess I am forgetting to do something or doing something wrong. May be there is a better option/action to do this! I'll appreciate any help you could give me! Thanks in advanced Helena Helena Garcia-Nieto de Mir Morodo Group VoIP Department E-mail: helena.gnieto(a)morodo.co.uk Morodo Limited voip(a)morodo.co.uk / www.mo-call.com Place of Registration: Companies House, England & Wales / Registered Office: 7 Breasy Place, 9 Burroughs Gardens Hendon, London NW4 4AU

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Managed ESL: AccessViolationException
by Grant Bagdasarian 19 May '14

19 May '14

Hello, I'm trying to run the following code, but after a few seconds I get AccessViolationException: class Program { static void Main(string[] args) { ESLconnection connection = new ESLconnection("192.168.1.1", 8021, "ClueCon"); connection.SendRecv("event plain ALL"); Task task1 = Task.Run(() => Receive(1, connection)); Task task2 = Task.Run(() => Receive(2, connection)); Console.ReadLine(); } static void Receive(int id, ESLconnection connection) { while (connection.Connected() == 1) { ESLevent eslEvent = connection.RecvEvent(); Console.WriteLine("{0} - {1}", id, eslEvent.Serialize("").Split(new char[] { '\r', '\n' })[0]); } } } What the basic idea behind this is, is to have a Task continuously listening for a CHANNEL_HANGUP event, and have the other Task process the call in parallel. Does the exception occur because multiple parallel tasks are trying to access the connection object? Is the only solution to create a second ESLconnection (HANGUP Handler) and register it to receive only calls for a given UUID? Regards, Grant

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Should I report a Lua problem in Freeswitch Jira?
by Alex Lake 19 May '14

19 May '14

Got a problem with a setup using FreeSWITCH Version 1.2.24+git~20140517T074940Z~a9a6e70915~64bit (git a9a6e70 2014-05-17 07:49:40Z 64bit) The problem is that on execution of certain Lua code (which uses luacurl), the freeswitch process terminates. The thing is, that I believe that this could be a memory leak or some funny exception in Lua, not really in Freeswitch. However, it does kill freeswitch, so might be of interest. Not knowing the architecture here, I'm not sure if it's anything the freeswitch community can do anything about! I'm happy to file the report, but only if it's not a complete waste of time ;-)

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How to hangup a call which is in a parked state
by Grant Bagdasarian 19 May '14

19 May '14

Hello, How do I hangup a call which is currently in a parked state? Documentation https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park says: - Set caller in a place where the channel won't be hungup on, while waiting for someone to talk to. Does this mean I can't hang up a channel that is in a parked state? I'm using the ESL library and I've tried different things: - ESLconnection Execute Hangup - SendMSG Hangup - Uuid_transfer to extension which hangs up the call None of them work. I also don't see the action back in the fs_cli output. The call was in a bridged state with another channel (inbound channel). After a few minutes I performed a uuid_transfer to park both calls, then uuid_broadcast to both calls to play a different prompt to each leg. Then I hangup the inbound channel by hanging up the phone, and the outbound channel remains in parked state. I have no idea how to force this channel to be hung up as well. Any ideas?

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How to use say_string
by mayamatakeshi 19 May '14

19 May '14

Hello, before I open a Jira ticket, as I might be overlooking something. I am trying to use say_string. Documentation is lacking so all we have is here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say But when I try to use it in the CLI i get: freeswitch@192.168.2.154@internal> eval ${say_string en.wav en current_date_time pronounced ${strepoch()}} file_string://time/mon-4.wav!digits/19.wav!digits/2.wav!digits/thousand.wav!digits/14.wav!time/at.wav!digits/1.wav!time/oclock.wav!time/p-m.wav As we can see, say_string is not generating full path to the files so when i try to use it with application playback for example I get "FILE NOT FOUND" response from FS. The say application itself works, so I suspect this is a bug, but since this behaviour happens with both latest head commit 34807208c672ce1afbd5d4f2984b5fddafe701c9 Author: Steve Underwood <steveu(a)coppice.org> Date: Sat May 17 14:28:58 2014 +0800 and a very old commit that I have in another machine: commit 1aa910339adb6f39df190137af439e34774908bd Author: Jeff Lenk <jeff(a)jefflenk.com> Date: Thu Mar 1 15:55:32 2012 -0600 I am inclined to think I am missing some configuration for this to work. Regards, Takeshi

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