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a problem about freeswitch conference caller control
by Erjian Li 01 Mar '12

01 Mar '12

Hi All, Firstly, I want to give my thanks to every one in the mailing list, especially the ones who helped me on my questions. My two questions about freeswitch has been solved due to the helps provided here. Today I meet a new question about freeswitch: My freeswitch server has been connected to a SIP provider's server (IDT server), and can dial out to normal cellphone numbers. Now I want to start a conference call using "conference <conf-name> dial ..." command, when the destination phones join the conference, the phone callers can't control the conference by pressing the keys specified in <caller-controls> section of conference.conf.xml. (In other words, if I press key '0', it can't mute myself; press '#', it can't hang up, etc. I use the default caller-controls group.) I want to know in this situation, whether the cellphone's key tone can't be transferred to freeswitch server? Can freeswitch server only receive DTMF encapsulated in RTP packet transferred over IP? I can control the conference when I use X-Lite as client, and freeswitch outputs following log when I press key '0' in X-Lite interface: ================================== freeswitch@eli-desktop> 2012-02-29 17:02:59.349272 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 0:960 2012-02-29 17:02:59.367880 [DEBUG] mod_conference.c:2919 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav' for play ================================= In the case of cellphone, when users press key, the DTMF is generated and transferred via PSTN channel to the IDT server, is this correct? and if so, is it up to IDT server to encapsulate the DTMF in RTP packets and send it to freeswitch server? -- Best Regards Erjian

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Subject: ClueCon 2012 - Call For Speaking Proposals
by Michael Collins 29 Feb '12

29 Feb '12

Greetings! The ClueCon team is gearing up for this summer's event, which will be held August 7-9, 2012 at the Wyndham in downtown Chicago. We want to hear your proposals for presentations to be given at ClueCon '12. Here's what we're looking for: - Discussions on technology or software that relate directly or indirectly to open source telephony - Working title or very brief description of the talk - A two or three sentence abstract giving more details - Speaker's name, company, and biographical information - All presentations should be 30 minutes in length, including question and answer time - ClueCon sponsors who wish to have a presentation always get first priority when it comes to scheduling Please send all proposals to marketing(a)cluecon.com. If you spoke at ClueCon last year and your bio has not changed then please make note of this fact. We look forward to hearing from you! The ClueCon Team http://www.cluecon.com

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FreeSWITCH Community Conference Call Today
by Michael Collins 29 Feb '12

29 Feb '12

Hey all! Don't forget our call in about 15 minutes. Agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_29 Talk to you soon! -Michael

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Mailing List Reminders
by Michael Collins 29 Feb '12

29 Feb '12

Hello All! It seems we've had a little drama on the mailing list (ML) lately so I thought I would take this opportunity to offer a few reminders... - This is a public list, so anything you say will be forever etched into the annals of Internet lore. Please think before you click Send. - This is a public list, so be aware that others may say things that are offensive to you personally but not to the subscribers in general. Please be ready to "deal with it" on occasion. - This list is moderated to a degree. We keep only a loose reign on the users here. However, if after several warnings a user repeatedly engages in trollish behavior then he or she WILL get moderated. Those who are incorrigibly trollish will be banned. I appreciate that most of our readers don't need these reminders, so thank you for indulging us. Keep up the good work! -Michael

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Write access permission to repository
by Erjan Tylepov 27 Feb '12

27 Feb '12

Hi all, I've generated rsa keys in openssh, sent a public one to the git-access(a)freeswitch.org yesterday, hadn't recieved any acknowledgement from then yet. Today I tried to: "git clone ssh://git@git.freeswitch.org:222/freeswitch.git" and got: "git(a)git.freeswitch.org's password:" What else I need to do to get write access? Regards, Yerzhan.

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Re: [Freeswitch-dev] Testing mod_sms
by Warren Lin 25 Feb '12

25 Feb '12

________________________________ From: Warren Lin <wlin121443(a)yahoo.com> To: "freeswitch-dev(a)lists.freeswitch.org" <freeswitch-dev(a)lists.freeswitch.org> Cc: Anthony Minessale <anthony.minessale(a)gmail.com> Sent: Saturday, February 25, 2012 11:58 AM Subject: Re: Testing mod_sms Hi, Anthony, Sorry for the late reply, I can only do the test over weekend... to re-cap, I was testing "a message was sent while target cell phone was off" , hopefully to get a notice when such non-delivery happens. Or someway , somehow to tell if a message was successfuly sent to target cell phone or not (not to the freeswitch). it looks like your patch did send out an event in this situation. but that causes another problem. freeswitch is trying to re-send the message again and again without any waiting in between....that took over whole CPU and hung my machine.... I would like to make following suggestion with my narrow view scope (I am using ESL) . see if it can be used to fit a broader scope. (1) mod_sms is mostly used by external programs such java client or php.... Is that possible let the application programer to decide to do the "re-send" or not. in the case of progammer decide to do the re-send, they should be able to define the re-send frequency and how many time it re-sends. At the end of re-trying. there will be an event to tell the re-try fail or not. Or make life easier, just leave the logic to the application programmer. "do not re-send at all". but do issue an Event. (this event needs some id, pls see #2 as follows.. (2) I could not examine the detail of the event you send back for the non-delivery ( since my machine hung, need to reboot. lost all the message on the console). but anyhow, in the event, hopefully there is some correlation-id. to help programmer to see which message was sent sucessfully which one is not.... in this case, there will be one id to return from two different places. one id return right on the call returnEvt=con.sendEvent(evt); the other is when you generates non-delivery event. so that they can be related. Hmm....I briefly traced the code. this implemention "might" need big surgery on existing code...(just maybe since I don't know too well about the code...) but least, on the non-delivery event, if you can keep ORIGINAL "to" "from" "event-timestamp" in the header. so a programmer can co-relate back the original msg. this is just my 2 cents. BTW, I am also looking into "clustering" freeswtich without using iptel.org. could you give me some heads-up. since I google this topic. but not many good solutions... Thanks Warren ________________________________ From: Anthony Minessale <anthony.minessale(a)gmail.com> To: Warren Lin <wlin121443(a)yahoo.com> Sent: Tuesday, February 21, 2012 11:01 AM Subject: Re: Testing mod_sms hi you can find out how to use the lists at http://lists.freeswitch.org try latest GIT, i put in a patch to fire the events with the heder Failed-Delivery true if you are listening for MESSAGE events you should catch them. On Mon, Feb 20, 2012 at 10:25 PM, Warren Lin <wlin121443(a)yahoo.com> wrote: > Hi Anthony, > > Found your e-mail address on the one of freeswitch discussion thread. do > not know how to post a msg there. > so, I send this mail. hopefully you can get this. > > I have used you mod_sms with ESL and Xlite, that works very well. but it > looks like > there is no way to get the message delivery confirmation back from > freeswitch. > > on this senario, a message were sent out while target cell phone was off. > > I used ESL in java to test this senario. I do see something from the log... > > .... > 2012-02-20 16:55:03.144870 [WARNING] sofia_presence.c:194 Not sending to > local > box for 2001(a)xxx.xxx.xxx > 2012-02-20 16:55:03.144870 [ERR] sofia_presence.c:243 Chat proto [global] > .... > ... > Nobody to send to: Profile internal > ... > but ESL can not see any "Event" for this non-delivery... Is there any way > to trap this event on ESL side? > > Warren > -- 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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FreeSWITCH Cookbook Is Published!
by Michael Collins 23 Feb '12

23 Feb '12

Hey all, Just wanted to mention a little story I wrote up on the main FreeSWITCH website: http://www.freeswitch.org/node/381 Packt has officially published the the cookbook, so get your copy today! -Michael

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my problems on originating a new call and saying digits
by Erjian Li 23 Feb '12

23 Feb '12

Hi All, I have a problem about he API command <originate>, so I'm here for some help. Thanks in advance. In FreeSWITCH, I want to originate a call to a remote user, when the user answers the call, FreeSWITCH will say some digits to the user. My command is as following: ======================================= freeswitch@internal>originate sofia/internal/8613302033450@xx.xx.xx.xx:5060 '&say(en number iterated 12345678912345)' ======================================== After I execute this command, the destination cellphone rings, but when I answer the call, the digit string "12345678912345" has almost finished saying, so I can't hear the complete digit string. In other words, the 'say' application has been executed before the call is answered. This is not what I expect. Could any one give me some advice on what I should do ? Or some other commands to meet my requirement? Any advice would be appreciated. BTW, if I originate a call to local user (e.g. 1005, which is the preconfigured user account of FreeSWITCH), and say some digits, the behavior is what I expect. For example, if I execute the below command: ======================================= freeswitch@internal>originate sofia/internal/1005%10.20.1.158 '&say(en number iterated 12345678912345)' ======================================= The digits "12345678912345" begin to be said right after I answer the call (I use X-Lite as the soft phone). This is what I expect. But why the behavior when with a remote user is different? -- Best Regards Erjian

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Look for some advice on how to implement a FreeSWITCH controlling module
by Erjian Li 22 Feb '12

22 Feb '12

Hi everyone, I am developing software for a conference call product with FreeSWITCH as IP PBX software, and I have some troubles on a module implementation. So I come here for some advice. Thanks in advance. My requirement of the module: I want to implement a controlling module (or separate program) for FreeSWITCH, this module (or separate program) receives the admin messages from a so-called admin server , parses the messages, and generates FreeSWITCH commands according to the message data, then send these commands to FreeSWITCH to execute (e.g. send a "conference dial ..." command to add a member into the conference call). Also, this module (or separate program) is required to listen the CHANNEL related events fired by FreeSWITCH, and report the FreeSWITCH channel status to admin server. My questions: I want to implement above functionality in a separate program. When running this program communicates with FreeSWITCH via FreeSWITCH's event socket library, is it applicable? I don't have much knowledge about FreeSWITCH's loadable module, is it better to implement it as a FreeSWITCH's loadable module, like the modules in src/mod/applications? -- Best Regards Erjian

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FreeSWITCH Conference Call Tomorrow (Wed Feb 22)
by Michael Collins 21 Feb '12

21 Feb '12

Hey all! Yes, I promise that I will come up for air and hang out with the community tomorrow. :) We have a special guest joining us tomorrow: Stefan Wintermeyer from Amooma. He will be talking about Gemeinschaft v4 and all the interesting things that lead to going with FreeSWITCH instead of Asterisk. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_22 Look forward to speaking with you all tomorrow! -Michael

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