- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Connect to SIP Trunking
by Roamer Roamer 02 Dec '13

02 Dec '13

Hi all, I'm new to freeswitch and want to learn SIP development, I have a question: what's protocol is using for the SIP PBX(freeswitch, asterisk, sipx) connect to SIP Trunking? In freeswitch source code, which part source code is for connect to SIP trunking ? Thank you in advance.

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PostgreSQL for windows build
by Ali R. 27 Nov '13

27 Nov '13

Hello All, How can I compile FreeSWITCH with PostgreSQL support with Visual Studio 2012. I understand under Linux you pass "--enable-core-pgsql-support" to the configure script. Many Thanks,

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Error handling / Error callbacks in Freeswitch
by Bhavesh 26 Nov '13

26 Nov '13

I want to know are there errorhandlers API available in freeswitch to which I can connect my method. e.g. in the event of error in freeswitch(e.g -ERR no reply), errorhandler set to my error handler function as callback say freeswitch_seterrorhandler(myerrorfunction) -- Bhavesh Soni Sr Software Engineer Great Software Laboratory Pvt Ltd 4th Floor, Pride Purple Accord, S.N. 3/6/1, Baner road, Pune - 411 045 (O) +91 20 4671 1584 (M) +91 86000 64681

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Is there any error handler in freeswitch
by Bhavesh 25 Nov '13

25 Nov '13

Freeswitch generally responds with " [ERR] " for any error condition. e.g. [ERR no reply] I want to know if there is any errorhandler (OR user defined error callback) methods implemented, so that in my freeswitch API if above kind of ERR occurs I can handle next procedure in my function. e.g. I want myErrHandler() function to print custom error message received from freeswitch ERR and give control to my function. -- Bhavesh Soni Sr Software Engineer Great Software Laboratory Pvt Ltd 4th Floor, Pride Purple Accord, S.N. 3/6/1, Baner road, Pune - 411 045 (O) +91 20 4671 1584 (M) +91 86000 64681

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Hangup hook gets destroyed when calling a script in conference caller controls
by Melanie Treitinger 18 Nov '13

18 Nov '13

Hi, I have the following scenario: When using a script in the conference caller controls, the hangup hook gets destroyed. Affected script languages: Lua, Spidermonkey, Python, Perl You can easily reproduce this effect using the below Lua scripts and adding the script application to the caller controls. This is what happens: - hangup without pressing a digit: hangup hook is called - press a digit without script call, then hangup: hangup hook is called - press a digit with script call, then hangup: hangup hook is not called Looking into the source code of session_destroy, you can see that the hangup hook is in fact removed - which shold not happen when calling a script. And there is no way to set it again after pressing the "*" digit. Any ideas on this? Thanks, Melanie test-conferece.lua: -------------- function session_hangup_hook(status) freeswitch.consoleLog("NOTICE", "Session hangup: " .. status .. "\n") error() end session:answer(); session:setHangupHook("session_hangup_hook") conf_num = "3000" session:execute("conference", string.format("%s@default", conf_num)) session:hangup() caller_controls.lua: -------------------- freeswitch.consoleLog("NOTICE","---- DTMF detect ----\n"); conference.conf.xml: -------------------- <caller-controls> <group name="default"> <control action="mute" digits="0"/> <control action="execute_application" digits="*" data="lua caller_controls.lua"/> <control action="energy up" digits="9"/> <control action="energy equ" digits="8"/> <control action="energy dn" digits="7"/> <control action="vol talk up" digits="3"/> <control action="vol talk zero" digits="2"/> <control action="vol talk dn" digits="1"/> <control action="vol listen up" digits="6"/> <control action="vol listen zero" digits="5"/> <control action="vol listen dn" digits="4"/> <control action="hangup" digits="#"/> </group>

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The gentones application generates a very bad quality sound
by Anton 17 Nov '13

17 Nov '13

I try to transfer the sound of 1,000 Hz from FreeSwitch to the browser and watch the quality of the sound, depending on the client type. Especially for this experiment, I created an external sip-profile: <gateway name="sip2sip.info"> <param name="username" value="soundtest"/> . . . </gateway> and following dialplan: <include> <extension name="soundtest(a)sip2sip.info"> <condition field="destination_number" expression="^soundtest$"> <action application="gentones" data="%(3500,0,1000)|-1"/> </condition> </extension> <extension name="mod_rtmp"> <condition field="source" expression="^mod_rtmp$"> <action application="gentones" data="%(3500,0,1000)|-1"/> </condition> </extension> </include> And reproducing clients: http://call2sip.ru - created on the basis of SIPml-api (correct work only under the Chrom) and http://soundtest.iiiii.info - created on the basis of mod_rtmp flex- client side Sound is produced as a result of the test recorded on the mic. At first, made the call from the SIPml-client to the number soundtest(a)sip2sip.info and recorded the resulting sound. Then made a call from the flex-client (in which case the destination number does not matter) and also recorded the resulting sound. Both sound recorded in a single file. Here's a link to this file: http://cs536109v4.vk.me/u78896495/audios/0efae0afdb31.mp3 As you can see this, copies of the same signal are quite different in quality. In other words, the sound reproduced through the flex contains a very noticeable defects. But it is only in the case where the sound is played in Windows 7. About other systems say anything until I can not. In this Friswitch was running on a remote server on CentOS 6.3. Yet I performed the same test when Friswitch was launched in the same system that plays sound. In this case, 50% of attempts were given a faulty sound, and the remaining 50% were given a normal tone. I also tried to listen to the audio tone through this service: http://s.phono.com/releases/1.1/samples/kitchen-sink/www/index.html And he also got a lot of noise. I would like to know whether it is possible to somehow get rid of this noise? Not giving up the use of flex-technology.

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FreeSWITCH 1.2.15 Released!!!
by Ken Rice 16 Nov '13

16 Nov '13

Greetings FreeSWITCHers, The FreeSWITCH Development Team is proud to announce the release of FreeSWITCH 1.2.15! Where can you find it? http://files.freeswitch.org/freeswitch-1.2.15.tar.bz2 If you are tracking the FreeSWITCH Yum or Deb repos, packages are available via normal upgrade mechanisms. Whats new you might ask? Not a lot, this is a maintenance release and includes many bug fixes. And don¹t forget to follow us on Twitter, and your other favorite social media! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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Re: [Freeswitch-dev] How to get value of the sample?
by Anton 15 Nov '13

15 Nov '13

I ran some tests, including one that led You. Even tried the following conversion types: char * samples = (char *) frame-> data; and anyway, the result is a -1 for all x, from 0 to frame->datalen-1 My task was to remake the module mod_avmd, ie file mod_avmd.c. And this problem has put me into a dead end, to the point where I moved my job to another server. In other operating environments do not have this problem. I wrote about this in more detail in the Issues: http://jira.freeswitch.org/browse/FS-5944

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How to get value of the sample?
by Anton 11 Nov '13

11 Nov '13

Hallo. Why, when I'm trying to get a sample of the sound, I always get the same value -1? For example, in the mod_avmd.c is the string: frame = switch_core_media_bug_get_read_replace_frame(bug); After this I add the following code switch_log_printf( SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "data[0]=%d\n", ((int16_t *)(frame->data))[0] ); Then trigger the module mod_avmd and get a result in the console: data[0]=-1; And so on for all the modules that contain the string: frame = switch_core_media_bug_get_read_replace_frame(bug); independently of the level of sound in the channel. Why is that?

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About the hardware fax detection on Sangoma cads
by sparklezou 11 Nov '13

11 Nov '13

Hi All, When configure the card and freetdm via wancfg_fs command, there is a question. Would you like to enable hardware fax detection? I always choose YES. I don't know how to use it. Is there any predefined dialplan context,such as "fax",to handle the incoming the fax? Thanks! 2013-11-11 sparklezou

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