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OSTAG - Open Source Telephony Action Group
by Michael Collins 03 Jun '10

03 Jun '10

Good news! The Open Source Telephony Action Group (OSTAG) is almost ready to become an official 501(c)3 non-profit organization. We need to raise $800 in order to file the necessary paperwork. We have set up a PayPal account where we can all send our donations: donations(a)ostag.org Please assist us in moving this important organization forward. OSTAG is dedicated to the advancement of open source telecommuncations software that improves the lives of people all over the world. This is a most worthy cause! Please help now and in the future. -Michael

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Status of Bridged Call
by Thirupathi 03 Jun '10

03 Jun '10

Hi, I am receiving the inbound call and i am bridging that call with some other number. But i need to know whether the bridged call (dial out call) has been ringing or answering. I am using the ESL connection method to communicate with the Freeswitch. Here is the snapshot of my code. Receiving inbound call : @sd = s @con = ESL::ESLconnection.new(s.to_i) @con.execute("linger") @con.setEventLock("true") @con.sendRecv("myevents") @con.execute("answer") ... .... After that i am playing and receiving input from the caller then i am bridging the call with some other number. Bridge Call: @con.execute("bridge", "sofia/gateway/junctionnetworks//#{@destination_number}") and waiting for the bridge event WaitForBridgeEvent: def waitForBridge while @con.connected e = @con.recvEvent() next if e == nil printEventData(e) handleHangup if hangupOccurred(e) if e.getHeader("Event-Name") == "CHANNEL_BRIDGE" && e.getHeader("Channel-State") == "CS_EXECUTE" answer_state = e.getHeader("Answer-State") if answer_state == "answered" call_answer_time = e.getHeader("answered_time") end_point_disp = e.getHeader("endpoint_disposition") var_end_point_disp = e.getHeader("var_endpoint_dispoisition") return call_answer_time, end_point_disp, var_end_point_disp end end end end In this code the answer_state is always returning the answered. Also there is always no value for call_answer_time, end_point_disp, var_end_point_disp. Bridging is happening and i have no problem with bridging the call. Please let me know whether the answer-state is returned in this case is for the inbound call or outbound call. (because i am wondering how it is always answered?). Also please help me to get the bridged call status (whether answered, ringing or others) Thanks and Regards, Thirupathi

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Freeswitch compilation and openldap
by François Legal 03 Jun '10

03 Jun '10

Hello, I'm wondering whether it would be possible to add a configure option for those (like me) who already have openldap and libldap installed and available on their systems, so that there would not need to have openldap downloaded and compiled during freeswitch compilation, but rather link to the available libldap ? François

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Help get freeswitch.com back
by Anthony Minessale 03 Jun '10

03 Jun '10

Some squatter has had freeswitch.com for years and it's time we had it back. We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party panel to resolve the dispute. They are using the domain to pose as a VoIP site (if you keep loading http://www.freeswitch.com/ you can see it in action) Don't let creeps like this misuse the internet! Use the paypal button on the site to donate to the cause. Everyone who donates will get their name up on our thank you page on our site. Put "freeswitch.com UDRP and your name as you would like to see it on the thank-you page" in the note on the paypal form. -- 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 <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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Small error in mod_freetdm
by François Legal 02 Jun '10

02 Jun '10

Hello, I recently upgraded to latest snapshot, and started using freetdm instead of openzap, and I found a small error in mod_freetdm. When setting up the channel name in function ftdm_channel_from_event line 1286, the line chanid = ftdm_channel_get_span_id(sigmsg->channel); should probably be replaced by chanid = ftdm_channel_get_number(sigmsg->channel); I wanted to post this on JIRA, but was not able to find a subproject for freetdm in JIRA. I also ported my patches from openzap to get the CID onto FXS channel work on european hardware with freetdm. Please let me know where I should post these. François

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Robust Affordable Speech Recognition
by Kashif Kahn 02 Jun '10

02 Jun '10

Dear All, All those who have wanted a speech recognition solution for Freeswitch but found the software cost too expensive or the recognition accuracy unsatisfactory, please consider Vestec Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license for the standard engine - is available for only $25. Additional ports (channels) licenses can be purchased for $99/port. The engine comes with a free-of-charge Freeswitch connector, thereby allowing direct interaction via Dialplan. Best regards, -Kashif Kashif Kahn VP, Business Development Vestec, Inc. Waterloo, ON Canada phone: (519) 885-7615

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FreeSWITCH Conference Call Starting Shortly! Kamailio being discussed today
by Michael Collins 02 Jun '10

02 Jun '10

Come on down: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_02 Daniel Constantin-Mierla (aka "miconda" or Twitter and IRC) is talking about Kamailio. -Michael

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Additional support for synchronization functions
by David Swardstrom 01 Jun '10

01 Jun '10

I got so involved in getting the proposed optional third parameter to setGlobalVariable() that I did not consider all of the possibilities. Using the change along with getGlobalVariable() is ok for data that would only change or for data that does not need to be completely unique. I also have a need to have a global counter that can both be incremented and decremented. This counter is designed to give me more control over the number of calls than the current mod_limit will provide. This would be used during a period where they may be several calls either entering the system, exiting the system, or both. It would be nice to have routines that would just increment or decrement safely. They would be much faster. I have spent the last couple of years working with Visual Studio and C++. The natural solution I would take in this situation would be to use InterlockedIncrement() and InterlockedDecrement(). In this environment I don't think I want to propose this. What I would like to propose is the following set of routines (using JavaScript notation): // Initialize the counter to a specified value initGlobalCounter(string name of variable, initial value); // Add a value to the global counter addGlobalCounter(string name of variable, value to add); // Subtract a value from the global counter reduceGlobalCounter(string name of variable, value to subtract); I am open to suggestions as to names of the routines. Even though all I need (now) is increment and decrement, since the code will need to use the Global Variable Mutex locks, it might as will be more general. Regards, Paul David Swardstrom

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Bridge Call Status
by Thirupathi 31 May '10

31 May '10

Hi, I am receiving the inbound call and i am bridging that call with some other number. But i need to know whether the bridged call (dial out call) has been ringing or answering. I am using the ESL connection method to communicate with the Freeswitch. Here is the snapshot of my code. Receiving inbound call : @sd = s @con = ESL::ESLconnection.new(s.to_i) @con.execute("linger") @con.setEventLock("true") @con.sendRecv("myevents") @con.execute("answer") ... .... After that i am playing and receiving input from the caller then i am bridging the call with some other number. Bridge Call: @con.execute("bridge", "sofia/gateway/junctionnetworks//#{@destination_number}") and waiting for the bridge event WaitForBridgeEvent: def waitForBridge while @con.connected e = @con.recvEvent() next if e == nil printEventData(e) handleHangup if hangupOccurred(e) if e.getHeader("Event-Name") == "CHANNEL_BRIDGE" && e.getHeader("Channel-State") == "CS_EXECUTE" answer_state = e.getHeader("Answer-State") if answer_state == "answered" call_answer_time = e.getHeader("answered_time") end_point_disp = e.getHeader("endpoint_disposition") var_end_point_disp = e.getHeader("var_endpoint_dispoisition") return call_answer_time, end_point_disp, var_end_point_disp end end end end In this code the answer_state is always returning the answered. Also there is always no value for call_answer_time, end_point_disp, var_end_point_disp. Bridging is happening and i have no problem with bridging the call. Please let me know whether the answer-state is returned in this case is for the inbound call or outbound call. (because i am wondering how it is always answered?). Also please help me to get the bridged call status (whether answered, ringing or others) Thanks and Regards, Thirupathi

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Record in Freeswitch
by Thirupathi 31 May '10

31 May '10

Hi, i am using record function to record the user information through channel into the voice file. @con.execute("record", "#{record_file_path} #{record_duration} #{silence_threshold}") I have specified the full path with the record_file_path variable. But i got the below error 2010-05-24 12:53:32.917541 [ERR] mod_sndfile.c:194 Error Opening File [/opt/freeswitch/sounds/123.wav] [System error : No such file or directory.] Is there any record function available in the freeswitch so that it will create the file automatically in the specified record_file_path parameter. Thanks and Regards, Thirupathi

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