- FreeSWITCH-users - FreeSWITCH Mailing Lists

Sending message
by Elliott Vogel 02 Feb '12

02 Feb '12

Is there a way to send an im to a sip user using the prompt?

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Good News Everybody - As Professor Hubert J Farnsworth would say!
by Ken Rice 02 Feb '12

02 Feb '12

Just a quick Update... BKW aka Brian West has made it out of the hospital in 1 piece... (except for the pieces the vampires^H^H^H^H^H^H^H errrrr doctors wanted to keep. He¹s currently recuperating from the comforts of his own home. Prognosis is good at this time, but we do know there are going to be follow up visits on this... If you havent already done so please consider tossing in a few bucks to help out with his hospital stay. As most of you in the US knows, insurance is not a total life saver for a week long stay with the Drs and Nurses... Send any donations to brian(a)bkw.org and please mark them personal and drop him a note to get well so paypal doesn¹t pull what paypal is known for... If you want to send something other then paypal I¹m sure he would appreciate suite cases of cash or precious metals. Additional contact information for such items can be found on http://bkw.org And to Brian, Thanks for all the years of helping people in the OpenSource VoIP community. I for one wouldn¹t be here today if it were not for your help all those years ago when I first discovered this type of software. Rest and Get Well my friend! K

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SIP Multiplexed Trunk as IAX2 or TDM
by Josue Diaz Cruz 02 Feb '12

02 Feb '12

Is there any way to have a SIP Trunk. I mean to have for example 32 channels merged in one? or something like this? When i try to find SIP trunk on internet i just see options for TDM gateway or similar but not really a multiplexed trunk. Can we do something with freeswitch? Josue Diaz Cruz Departamento Tecnico y Soporte <mailto:jdiaz@coinfru.com> jdiaz(a)coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia <http://www.coinfru.com/> www.coinfru.com

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Method of checking if media is being played at a session
by Michael Lutz 02 Feb '12

02 Feb '12

Hi All, I am dealing with a little problem... To play asynchronous audio in my Lua script I use the api function "uuid_broadcast {uuid} {fileName} aleg". This all works well and audio is being buffered if I call my function multiple times. When I want to abort all plays and flush the buffer I use "uuid_break {uuid} all". This all works excellent! My problem is I at certain points need to know if audio is being played (or awaits in the buffer) and need to pause execution untill all audio is beging played (or flushed). Is there a way of doing this in either Lua or apiFunctions? or a channel variable perhaps? I have looked and tryed but cannot seem to find a way doing this. Any hints (or solution ;-) appreciated! Regards, Michael Lutz

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Community Alert: Our Friend Brian West Is In The Hospital
by Michael Collins 02 Feb '12

02 Feb '12

Hello FreeSWITCH Community, We wanted to let everyone know that Brian West is in St. Francis Hospital, Tulsa, OK. He is checked into room 4325. At this point it looks like he will be there for at least a week. I think it's safe to say that Brian is a dynamo not just for FreeSWITCH but for open source VoIP and telephony. My guess is that there's not a person who's worked with Asterisk or FreeSWITCH in the past 10 years who hasn't been aided in one way or another by Brian. Personally I can say that in the past five years Brian has helped me immeasurably. I think I have literally asked him thousands of questions and he has patiently answered them all. We would like to have everyone keep Brian in their thoughts and prayers. Please feel free to send him cards and well wishes at the hospital. Also, if you wish to send gifts and such to his home address you may do so: Brian West 714 Osage Ave McAlester, OK 74501-6638 For those who are interested in helping in more practical ways, such as helping to defray the considerable medical costs that no doubt will be incurred, we will have more information shortly. Please spread the word via Facebook, Twitter, and any other medium you have. Let's all let Brian know just how much we value him as a colleague and friend! -Michael Collins

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FreeSwitch compilation error...
by Kevin White 01 Feb '12

01 Feb '12

Hi all, Trying to get freeswitch to compile here. I took the following steps: 1. used the instructions to download the source from git 2. ./bootstrap.sh 3. Edit modules.conf to uncomment the flite and freetdm modules 4. ./configure 5. Make I get the following during the 'make' process: making all mod_dialplan_xml Compiling /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c... mkdir .libs Compiling /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c ... /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c: In function 'parse_exten': /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c:147: error: expected ';' before '}' token make[5]: *** [mod_dialplan_xml.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_dialplan_xml-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Any tips? Thanks, Kevin ________________________________ Kevin White Senior Software Engineer (801) 727-2362 office CentralLogic www.centrallogic.com<http://www.centrallogic.com>

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exit lua script in hangup hook
by Gerald Weber 01 Feb '12

01 Feb '12

Hello, i'm trying to exit my lua script (called from dialplan) to exit in the the hanguphook: function on_hangup(s,status) freeswitch.consoleLog("NOTICE","---- on_hangup: "..status.."\n"); error(); end freeswitch.consoleLog("NOTICE","---- ANSWER:\n"); session:answer(); freeswitch.consoleLog("NOTICE","---- SETHOOK: \n"); session:setHangupHook("on_hangup"); while (session:ready() == true) do freeswitch.consoleLog("NOTICE","---- START OF LOOP \n"); freeswitch.consoleLog("NOTICE","---- STREAMFILE \n"); session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-on_hold_indefinitely.wav"); freeswitch.consoleLog("NOTICE","---- END OF SCRIPT 1 -> HANGUP\n"); end freeswitch.consoleLog("NOTICE","---- END OF SCRIPT 2 -> HANGUP\n"); session:hangup() When I hangup during the streamFile Call, I geht the following output: 2012-01-25 12:10:19.936097 [NOTICE] switch_channel.c:930 New Channel sofia/internal/2001(a)192.168.20.73 [2b1cd9f6-4745-11e1-8c71-0d2938abf0b5] 2012-01-25 12:10:19.936097 [INFO] mod_dialplan_xml.c:481 Processing B#-FS-2001 <2001>->3001 in context default 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- ANSWER: 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:599 Channel [sofia/internal/2001(a)192.168.20.73] has been answered 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- SETHOOK: 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- START OF LOOP 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- STREAMFILE 2012-01-25 12:10:21.116050 [NOTICE] sofia.c:624 Hangup sofia/internal/2001(a)192.168.20.73 [CS_EXECUTE] [NORMAL_CLEARING] 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- on_hangup: hangup 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- END OF SCRIPT 1 -> HANGUP 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- END OF SCRIPT 2 -> HANGUP 2012-01-25 12:10:21.116050 [NOTICE] switch_core_session.c:1398 Session 2 (sofia/internal/2001(a)192.168.20.73) Ended 2012-01-25 12:10:21.116050 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/2001(a)192.168.20.73 [CS_DESTROY] Why does the hangup handler return and the rest of the script is executed ? I event tried "do return end;" , "exit;" ," exit();" instead of "error(); - none of them works. If i put the same logic into javascript, the hook works and the script is terminated: function on_hangup(s,status) { console_log("NOTICE","---- hangup "+status+"\n"); return "exit"; } console_log("NOTICE","---- ANSWER\n"); session.answer(); console_log("NOTICE","---- ANSWER\n"); session.setHangupHook(on_hangup); while(session.ready()) { console_log("NOTICE","---- START OF LOOP\n"); session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-on_hold_indefinitely.wav"); console_log("NOTICE","---- END OF LOOP\n"); } console_log("NOTICE","---- END OF SCRIPT\n"); session.hangup(); 2012-01-25 12:13:08.396041 [NOTICE] switch_channel.c:930 New Channel sofia/internal/2001(a)192.168.20.73 [8f85a15c-4745-11e1-8c76-0d2938abf0b5] 2012-01-25 12:13:08.396041 [INFO] mod_dialplan_xml.c:481 Processing B#-FS-2001 <2001>->3002 in context default 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- ANSWER 2012-01-25 12:13:08.396041 [NOTICE] mod_spidermonkey.c:2068 Channel [sofia/internal/2001(a)192.168.20.73] has been answered 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- ANSWER 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- START OF LOOP 2012-01-25 12:13:09.875996 [NOTICE] sofia.c:624 Hangup sofia/internal/2001(a)192.168.20.73 [CS_EXECUTE] [NORMAL_CLEARING] 2012-01-25 12:13:09.875996 [NOTICE] hangup.js:2 ---- hangup hangup 2012-01-25 12:13:09.875996 [NOTICE] switch_core_session.c:1398 Session 3 (sofia/internal/2001(a)192.168.20.73) Ended 2012-01-25 12:13:09.875996 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/2001(a)192.168.20.73 [CS_DESTROY] Any suggestions ? Or do i miss something here ? The dialplan part: <extension name="hanguptest"> <condition field="destination_number" expression="^(3001)$"> <action application="set" data="session_in_hangup_hook=true"/> <action application="lua" data="hangup.lua"/> </condition> </extension> (replace hangup.lua with hangup.js for the javascript version) Freeswitch Version: FreeSWITCH Version 1.0.head (git-9be51d5 2012-01-21 13-45-21 -0500) thx&regards, gw

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Re: [Freeswitch-users] Using Oracle as a database
by Gerald Weber 01 Feb '12

01 Feb '12

FS "pings " the db to keep the connection open. The statement is "select 1" by default, which is invalid for ORACLE. There is no config file to set this, you have to edit the file src/switch_odbc.c (iirc) and recompile. Sent from my Android phone using TouchDown (www.nitrodesk.com) -----Original Message----- From: Ira Tessler [ira(a)connectmevoice.com] Received: Mittwoch, 01 Feb. 2012, 20:44 To: FreeSWITCH-users(a)lists.freeswitch.org [FreeSWITCH-users(a)lists.freeswitch.org] Subject: [Freeswitch-users] Using Oracle as a database I have odbc set up for Oracle on my FS Linux box. I can connect to Oracle using isql and run queies. I am trying to configure Freeswitch to using the database but when I start up Freeswitch I get the following error message: 2012-02-01 14:29:07.056793 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN dbnj [STATE: HY000 CODE 923 ERROR: [Oracle][ODBC][Ora]ORA-00923: FROM keyword not found where expected Anyone have any ideas ? Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com<http://www.connectmevoice.com/>

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Using Oracle as a database
by Ira Tessler 01 Feb '12

01 Feb '12

I have odbc set up for Oracle on my FS Linux box. I can connect to Oracle using isql and run queies. I am trying to configure Freeswitch to using the database but when I start up Freeswitch I get the following error message: 2012-02-01 14:29:07.056793 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN dbnj [STATE: HY000 CODE 923 ERROR: [Oracle][ODBC][Ora]ORA-00923: FROM keyword not found where expected Anyone have any ideas ? Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com

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Don't want to display ringing BLF
by Rob Hutton 01 Feb '12

01 Feb '12

We have a group of 5 phones currently configured in a ring group that all have BLF keys monitoring each other. The problem is that when a call comes in, all of the BLF lines flash and no one can tell who is actually on the phone. Can anyone think of a configuration where all five phone can ring on an incoming call, they can see if the other phones are in use, and ringing is ignored? Thanks, Rob

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