From georgi_mei at abv.bg Tue Apr 2 11:25:57 2013 From: georgi_mei at abv.bg (Georgi Stefanov) Date: Tue, 2 Apr 2013 10:25:57 +0300 (EEST) Subject: [Freeswitch-dev] Development Info Message-ID: <402200569.84847.1364887557214.JavaMail.apache@mail21.abv.bg> Hello All, It is spring time (almost) again and as most of the things I need to reproduce or simply develop other kind of endpoint. The purpose: 1.I need other kind of signaling, a protocol which is not supported by FS 2.The need is the call to enter into the dialplan and execute it (FS to execute the dialplan a call from other signalization point is received) What I have done till now: I started from http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Before_You_Begin... I have used mod_sofia and mod_skypopen as reference as result I am (the call is) into the dialplan now (using switch_core_session_execute_exten) but I can not deal properly with channel state callbacks and no RTP data is flowing/traveling/sending. I have been given some hints to check-out rtp.c but now I think I need more info about creating channels and dealing with their states. Then I will need to deal with the RTP data So if anyone have some clues and wants to share...I will be very glad From marketing at cluecon.com Tue Apr 2 18:44:55 2013 From: marketing at cluecon.com (Michael Collins) Date: Tue, 2 Apr 2013 07:44:55 -0700 Subject: [Freeswitch-dev] ClueCon 2013 - Call For Speakers Message-ID: [image: https://mail-attachment.googleusercontent.com/attachment/u/0/?ui=2&ik=43fec29535&view=att&th=13cf45a70d222cc4&attid=0.1&disp=inline&realattid=f_hddkcyjh0&safe=1&zw&saduie=AG9B_P-p4MyvJI_ZqTegUZpanklK&sadet=1361819341430&sads=tdlIrK0Sx3z615AVwvfWzTRj7Bw] ClueCon - the open source IP communications conference by developers, fordevelopers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea fora technical presentation for ClueCon 2013 then we would like to hear about it. 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 - 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 WebRTC-related talks and demonstrations Please send your proposals to marketing at cluecon.com. Be sure to include the following items: - Working title - Brief description of the talk (abstract) - Name of the presenter 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. ClueCon 2013 Registration Information ClueCon 2013 registration is now open!. Visit the registration page for details. Be sure to book your room at the Hyatt Chicago Magnificent Mileand qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2013. Also, keep in mind that the FreeSWITCH community has a conference calleach Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2013. Stay tuned for more news about ClueCon speakers, sponsors, and related events! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130402/0a5faa63/attachment-0001.html From steveayre at gmail.com Tue Apr 2 22:15:19 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 2 Apr 2013 19:15:19 +0100 Subject: [Freeswitch-dev] Development Info In-Reply-To: <402200569.84847.1364887557214.JavaMail.apache@mail21.abv.bg> References: <402200569.84847.1364887557214.JavaMail.apache@mail21.abv.bg> Message-ID: docs.freeswitch.org is also a very useful reference On 2 April 2013 08:25, Georgi Stefanov wrote: > Hello All, > It is spring time (almost) again and as most of the things I need to > reproduce or simply develop other kind of endpoint. > > The purpose: > > 1.I need other kind of signaling, a protocol which is not supported by FS > 2.The need is the call to enter into the dialplan and execute it (FS to > execute the dialplan a call from other signalization point is received) > > What I have done till now: > > I started from > http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Before_You_Begin. > .. > I have used mod_sofia and mod_skypopen as reference > as result I am (the call is) into the dialplan now (using > switch_core_session_execute_exten) but I can not deal properly with channel > state callbacks and no RTP data is flowing/traveling/sending. > I have been given some hints to check-out rtp.c but now I think I need > more info about creating channels and dealing with their states. Then I > will need to deal with the RTP data > So if anyone have some clues and wants to share...I will be very glad > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130402/31d43f8d/attachment.html From ml-ktk at netlabs.org Wed Apr 3 19:34:49 2013 From: ml-ktk at netlabs.org (Adrian Gschwend) Date: Wed, 03 Apr 2013 17:34:49 +0200 Subject: [Freeswitch-dev] Compilation fails on FreeBSD Message-ID: <515C4C19.4040400@netlabs.org> Hi group, git master does not compile on FreeBSD right now, did a clean checkout followed by bootstrap & configure: ... quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -Ilibs/sofia-sip/libsofia-sip-ua/sdp -Ilibs/sofia-sip/libsofia-sip-ua/su -I/usr/local/include -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/local/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -I/usr/local/include -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_rtp.lo -MD -MP -MF .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_rtp.o cc1: warnings being treated as errors /usr/local/src/freeswitch/libs/srtp/crypto/include/datatypes.h:412: warning: 'be64_to_cpu' defined but not used gmake[1]: *** [libfreeswitch thanks Adrian From msc at freeswitch.org Wed Apr 3 21:10:52 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 3 Apr 2013 10:10:52 -0700 Subject: [Freeswitch-dev] Come join the conference call! Message-ID: We have several items to discuss, so please join us! http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_03 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130403/36b225c1/attachment.html From krice at freeswitch.org Wed Apr 3 23:44:15 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 03 Apr 2013 14:44:15 -0500 Subject: [Freeswitch-dev] Compilation fails on FreeBSD In-Reply-To: <515C4C19.4040400@netlabs.org> Message-ID: Open a jira please On 4/3/13 10:34 AM, "Adrian Gschwend" wrote: > Hi group, > > git master does not compile on FreeBSD right now, did a clean checkout > followed by bootstrap & configure: > > ... > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include > -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src > -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -Ilibs/sofia-sip/libsofia-sip-ua/sdp -Ilibs/sofia-sip/libsofia-sip-ua/su > -I/usr/local/include -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/libtpl-1.5/src > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/local/include > -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src > -I/usr/local/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -I/usr/local/include -DHAVE_OPENSSL -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_rtp.lo > -MD -MP -MF .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c > -fPIC -DPIC -o .libs/libfreeswitch_la-switch_rtp.o > cc1: warnings being treated as errors > /usr/local/src/freeswitch/libs/srtp/crypto/include/datatypes.h:412: > warning: 'be64_to_cpu' defined but not used > gmake[1]: *** [libfreeswitch > > thanks > > Adrian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From georgi_mei at abv.bg Thu Apr 4 11:54:26 2013 From: georgi_mei at abv.bg (Georgi Stefanov) Date: Thu, 4 Apr 2013 10:54:26 +0300 (EEST) Subject: [Freeswitch-dev] Development Info Message-ID: <1991079597.167.1365062066012.JavaMail.apache@mail21.abv.bg> Thanks Steven, I am looking the DOCs too, Currently I succeeded to place my non-sip call into the dialplan. I see the dialplan execution is started but when playback execution is started is stops with hanging up my call. ( I have tried to pre-answer but...) I need something like " your_io_routines.outgoing_channel is called create a new session (with switch_core_session_request()) parse outbound_profile->destination_number to get called number qualify the session, attached channel and attached private data The channel is currently in CS_NEW Change channel state to CS_INIT Make the phone ring (and call switch_channel_mark_ring_ready()) your_state_handler.on_init is called Change channel state to CS_ROUTING your_state_handler.on_routing is called What to do here? The call is answered switch_channel_mark_answered(channel) " the info from Authoring_Freeswitch_Modules but with more details In my case: I have no calling of your_io_routines.outgoing_channel I do the codec staff in on_init function On_Routing I do the pre-answer On_Receiving I try to start rtp session I can not understand am I doing it right or not ? (probably not, but I need to do it the right way, but I do not know the right way) So every kind of info is welcome ----------------------------------------------------------------- ?????? ????????? ?? ???-??????????????? ?? ???? ?? ???? ? ????????! http://www.carmarket.bg/ From msc at freeswitch.org Tue Apr 9 01:33:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 8 Apr 2013 14:33:12 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes Message-ID: A belated happy Monday to all! Last week on our conference callwe spent some time discussing various topics such as Jira system improvements and updates, sound prompts, ClueCon 2013 and several other topics. The recordings can be found in the usual place. On this week's call we are going to have a community discussion and a few minutes of Q&A from the audience. If possible, please add your questions to the agenda page and we'll research them prior to the call. Where applicable we'll ask members of the audience to update the wiki to reflect any undocumented knowledge that has been discussed. Things have been busy with the advent of ClueCon 2013season but we're on top of things and we'll keep everyone posted on all the particulars. Feel free to register at any time. Also, please contact us at this email address if you have any questions about being a speaker, sponsor, or attendee. We'll be glad to assist. One last item: I wanted to personally say thank you to Steven Ayre for all of his hard work with answering questions on the mailing list. He has done a great job of helping lots of people with a variety of questions. Many thanks to Steven and all the others who make FreeSWITCH such a great FOSS community. Thanks and have a great week! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130408/8c234c93/attachment.html From gvvsubhashkumar at gmail.com Wed Apr 10 11:32:55 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Wed, 10 Apr 2013 13:02:55 +0530 Subject: [Freeswitch-dev] Sound Not Ok(Choppy Sound) after 250 ports Message-ID: Hi, I ran the load test, raising the port count slowly (every 5 to 10 minutes or so) to 100, 150, 200, 250, 275 and 300 ports. There were no audio problems until I reached 275 ports. After probably 10 minutes at 275 ports, audio problems started, and after that they were consistent. Of course I can?t guarantee that there were no audio problems at lower port counts - if the problem only occurred intermittently I might not have caught it or I might not have waited long enough. Once the audio problems started, I slowly lowered the number of ports back to 250 ports. The audio problems disappeared after about 2 1/2 minutes at 250 ports. I then increased the number of ports back to 275 ports. Thanks, Subhash. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130410/46a35fab/attachment-0001.html From anthony.minessale at gmail.com Wed Apr 10 20:33:46 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 10 Apr 2013 11:33:46 -0500 Subject: [Freeswitch-dev] Sound Not Ok(Choppy Sound) after 250 ports In-Reply-To: References: Message-ID: Each physical setup has its own limitations. It's relevant to the hardware and software configurations on the box. Once you find that value for your particular machine, you can set the max sessions to that number. On Wed, Apr 10, 2013 at 2:32 AM, Subhash wrote: > Hi, > I ran the load test, raising the port count slowly (every 5 to 10 minutes > or so) to 100, 150, 200, 250, 275 and 300 ports. > > There were no audio problems until I reached 275 ports. After probably 10 > minutes at 275 ports, audio problems started, and after that they were > consistent. > Of course I can?t guarantee that there were no audio problems at lower > port counts - if the problem only occurred intermittently I might not have > caught it or I might not have waited long enough. > > Once the audio problems started, I slowly lowered the number of ports back > to 250 ports. The audio problems disappeared after about 2 1/2 minutes at > 250 ports. > I then increased the number of ports back to 275 ports. > > Thanks, > Subhash. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130410/0e810c60/attachment.html From msc at freeswitch.org Wed Apr 10 20:43:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 10 Apr 2013 09:43:50 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Hello all! Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 We have a few items to discuss, and we'll be asking the community for input on an important question: How do you choose whether to do closed, freemium, or open source when you've built a great solution? Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130410/1fde6624/attachment.html From steveayre at gmail.com Wed Apr 10 21:17:22 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 10 Apr 2013 18:17:22 +0100 Subject: [Freeswitch-dev] Sound Not Ok(Choppy Sound) after 250 ports In-Reply-To: References: Message-ID: Not to mention network bandwidth too. On 10 April 2013 17:33, Anthony Minessale wrote: > Each physical setup has its own limitations. It's relevant to the > hardware and software configurations on the box. > > Once you find that value for your particular machine, you can set the max > sessions to that number. > > > > On Wed, Apr 10, 2013 at 2:32 AM, Subhash wrote: > >> Hi, >> I ran the load test, raising the port count slowly (every 5 to 10 minutes >> or so) to 100, 150, 200, 250, 275 and 300 ports. >> >> There were no audio problems until I reached 275 ports. After probably 10 >> minutes at 275 ports, audio problems started, and after that they were >> consistent. >> Of course I can?t guarantee that there were no audio problems at lower >> port counts - if the problem only occurred intermittently I might not have >> caught it or I might not have waited long enough. >> >> Once the audio problems started, I slowly lowered the number of ports >> back to 250 ports. The audio problems disappeared after about 2 1/2 minutes >> at 250 ports. >> I then increased the number of ports back to 275 ports. >> >> Thanks, >> Subhash. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130410/a05962c3/attachment.html From gvvsubhashkumar at gmail.com Fri Apr 12 10:41:30 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Fri, 12 Apr 2013 12:11:30 +0530 Subject: [Freeswitch-dev] Sound Not Ok(Choppy Sound) after 250 ports In-Reply-To: References: Message-ID: Thanks, Can you please explain in detail what is that value and according to that how can i set the max sessions in freeswitch.How can i find the call limit of freeswitch on a particular OS.Do we have any formula to find the max call limit. Thanks, Subhash. On Wed, Apr 10, 2013 at 10:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Each physical setup has its own limitations. It's relevant to the > hardware and software configurations on the box. > > Once you find that value for your particular machine, you can set the max > sessions to that number. > > > > On Wed, Apr 10, 2013 at 2:32 AM, Subhash wrote: > >> Hi, >> I ran the load test, raising the port count slowly (every 5 to 10 minutes >> or so) to 100, 150, 200, 250, 275 and 300 ports. >> >> There were no audio problems until I reached 275 ports. After probably 10 >> minutes at 275 ports, audio problems started, and after that they were >> consistent. >> Of course I can?t guarantee that there were no audio problems at lower >> port counts - if the problem only occurred intermittently I might not have >> caught it or I might not have waited long enough. >> >> Once the audio problems started, I slowly lowered the number of ports >> back to 250 ports. The audio problems disappeared after about 2 1/2 minutes >> at 250 ports. >> I then increased the number of ports back to 275 ports. >> >> Thanks, >> Subhash. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130412/0c213385/attachment-0001.html From steveayre at gmail.com Fri Apr 12 11:56:01 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 12 Apr 2013 08:56:01 +0100 Subject: [Freeswitch-dev] Sound Not Ok(Choppy Sound) after 250 ports In-Reply-To: References: Message-ID: That's a very hard question to answer. There's a large number of factors that come in, not just the number of calls. CPU gets limited by what you're doing on the calls as well as just the number of them. Things like transcoding will increase the CPU usage per call and therefore reduce the number of calls you can handle. The amount of CPU used will vary between codecs. Conferencing, recording, voicemail etc too. Disk I/O might become an issue if you're doing a lot of recording or too much logging. And there are a number of factors about your OS that you can tune. Moving the core DB to either a ramdisk, ODBC or PgSQL can also have an effect. In short you need to load-test your specific use case since it may vary from what others are doing. Try starting here: http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations The easier limit to calculate is network bandwidth. That's related which codec you use and and the ptime you use (higher ptime = more fits in a single packets, so less overhead but you lose more if packets get lost). Try http://www.bandcalc.com/ to start with (though it doesn't list many codecs). If you're hearing the sound cutting in/out it could be packet loss rather than the server if the network isn't able to keep up. Use wireshark to capture the RTP stream and analyse it, look for packet loss or jitter. -Steve On 12 April 2013 07:41, Subhash wrote: > Thanks, > > Can you please explain in detail what is that value and according to that > how can i set the max sessions in freeswitch.How can i find the call limit > of freeswitch on a particular OS.Do we have any formula to find the max > call limit. > > > > Thanks, > Subhash. > > > On Wed, Apr 10, 2013 at 10:03 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Each physical setup has its own limitations. It's relevant to the >> hardware and software configurations on the box. >> >> Once you find that value for your particular machine, you can set the max >> sessions to that number. >> >> >> >> On Wed, Apr 10, 2013 at 2:32 AM, Subhash wrote: >> >>> Hi, >>> I ran the load test, raising the port count slowly (every 5 to 10 >>> minutes or so) to 100, 150, 200, 250, 275 and 300 ports. >>> >>> There were no audio problems until I reached 275 ports. After probably >>> 10 minutes at 275 ports, audio problems started, and after that they were >>> consistent. >>> Of course I can?t guarantee that there were no audio problems at lower >>> port counts - if the problem only occurred intermittently I might not have >>> caught it or I might not have waited long enough. >>> >>> Once the audio problems started, I slowly lowered the number of ports >>> back to 250 ports. The audio problems disappeared after about 2 1/2 minutes >>> at 250 ports. >>> I then increased the number of ports back to 275 ports. >>> >>> Thanks, >>> Subhash. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130412/7263f2e8/attachment.html From msc at freeswitch.org Fri Apr 12 22:29:42 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 12 Apr 2013 11:29:42 -0700 Subject: [Freeswitch-dev] Sound Not Ok(Choppy Sound) after 250 ports In-Reply-To: References: Message-ID: It sounds like you've already found that magic value. Like Steven says, it's very difficult to have a pure formula because of all of the factors that are beyond the control of the FreeSWITCH software. However, you're doing one thing quite correctly: putting real calls on a real server and listening to them. If there is a formula it must include having a knowledgeable person sitting down and actually listening to the calls to hear what happens when the audio starts to go bad. Steven mentioned some other good points as well. First off, you need to make sure that you aren't experiencing any packet loss. He mentioned Wireshark specifically. I'd like to suggest that you add pcapsipdumpto your toolbox. It grabs signaling and audio for an individual phone call and puts it all in a single file on disk. That makes it much easier to pick out a specific phone call and analyze it in Wireshark. Some other things you can try: run top command and see what your CPU usage is. Get iotop and see if there's anything issues with information flowing in/out NICs or to/from HDDs. Also, get another server and use faster CPU, more RAM, faster HDDs, etc. and see if your magic number changes on that server. I'm afraid that there's no simple formula. Of course, if there was then we'd all be out of a job. :) -Michael On Thu, Apr 11, 2013 at 11:41 PM, Subhash wrote: > Thanks, > > Can you please explain in detail what is that value and according to that > how can i set the max sessions in freeswitch.How can i find the call limit > of freeswitch on a particular OS.Do we have any formula to find the max > call limit. > > > > Thanks, > Subhash. > > > On Wed, Apr 10, 2013 at 10:03 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Each physical setup has its own limitations. It's relevant to the >> hardware and software configurations on the box. >> >> Once you find that value for your particular machine, you can set the max >> sessions to that number. >> >> >> >> On Wed, Apr 10, 2013 at 2:32 AM, Subhash wrote: >> >>> Hi, >>> I ran the load test, raising the port count slowly (every 5 to 10 >>> minutes or so) to 100, 150, 200, 250, 275 and 300 ports. >>> >>> There were no audio problems until I reached 275 ports. After probably >>> 10 minutes at 275 ports, audio problems started, and after that they were >>> consistent. >>> Of course I can?t guarantee that there were no audio problems at lower >>> port counts - if the problem only occurred intermittently I might not have >>> caught it or I might not have waited long enough. >>> >>> Once the audio problems started, I slowly lowered the number of ports >>> back to 250 ports. The audio problems disappeared after about 2 1/2 minutes >>> at 250 ports. >>> I then increased the number of ports back to 275 ports. >>> >>> Thanks, >>> Subhash. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130412/b3fb52b7/attachment-0001.html From jmesquita at freeswitch.org Mon Apr 15 01:59:01 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 14 Apr 2013 18:59:01 -0300 Subject: [Freeswitch-dev] Portaudio AEC take 2 Message-ID: Guys, I've been looking at an AEC solution to integrate on portaudio for the softphones to take advantage of it and it seems like WebRTC has one that could be used. I am not by any means an expert on the matter so before I start hacking things, I would like to ask your opinion. First of all, licensing... Is this license compatible? http://www.webrtc.org/license-rights/license If the license is compatible, where would be the best place to implement it? I would definitely need some pointers.. I found out that Blink is using it on PJMEDIA as this patch ( http://sipsimpleclient.org/projects/sipsimpleclient/repository/entry/patches/pjsip-webrtc_aec.patch) shows. PJMedia already has an interface for noise supression and AEC so I guess it was "easy" to implement there. We would implement it as a media bug or on the portaudio module itself? Also, if anyone knows if this will bring good results, would be great! AEC that just works is something that is really wanted on a softphone... Jo?o Mesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130414/18780d9a/attachment.html From mike at jerris.com Mon Apr 15 16:17:39 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 15 Apr 2013 08:17:39 -0400 Subject: [Freeswitch-dev] Portaudio AEC take 2 In-Reply-To: References: Message-ID: I'm working on this already. Mike On Apr 14, 2013, at 5:59 PM, Jo?o Mesquita wrote: > Guys, > > I've been looking at an AEC solution to integrate on portaudio for the softphones to take advantage of it and it seems like WebRTC has one that could be used. I am not by any means an expert on the matter so before I start hacking things, I would like to ask your opinion. > > First of all, licensing... Is this license compatible? http://www.webrtc.org/license-rights/license > > If the license is compatible, where would be the best place to implement it? I would definitely need some pointers.. I found out that Blink is using it on PJMEDIA as this patch (http://sipsimpleclient.org/projects/sipsimpleclient/repository/entry/patches/pjsip-webrtc_aec.patch) shows. PJMedia already has an interface for noise supression and AEC so I guess it was "easy" to implement there. We would implement it as a media bug or on the portaudio module itself? > > Also, if anyone knows if this will bring good results, would be great! AEC that just works is something that is really wanted on a softphone... > > > Jo?o Mesquita > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130415/33ae8440/attachment.html From steveu at coppice.org Mon Apr 15 17:14:17 2013 From: steveu at coppice.org (Steve Underwood) Date: Mon, 15 Apr 2013 21:14:17 +0800 Subject: [Freeswitch-dev] Portaudio AEC take 2 In-Reply-To: References: Message-ID: <516BFD29.8010402@coppice.org> Hi, Oh good. This has been on my todo list for a while. Its nice if someone saves me the effort. :-) The WebRTC canceller is the right one to use. Nothing else I've found is worth considering. The WebRTC canceller is the only open source one which allows for the mic and speaker sampling rates not matching. If you don't do that you should be OK on an embedded board or a phone, but you would be wasting your time on a PC. Regards, Steve On 04/15/2013 08:17 PM, Michael Jerris wrote: > I'm working on this already. > > Mike > > On Apr 14, 2013, at 5:59 PM, Jo?o Mesquita > wrote: > >> Guys, >> >> I've been looking at an AEC solution to integrate on portaudio for >> the softphones to take advantage of it and it seems like WebRTC has >> one that could be used. I am not by any means an expert on the matter >> so before I start hacking things, I would like to ask your opinion. >> >> First of all, licensing... Is this license compatible? >> http://www.webrtc.org/license-rights/license >> >> If the license is compatible, where would be the best place to >> implement it? I would definitely need some pointers.. I found out >> that Blink is using it on PJMEDIA as this patch >> (http://sipsimpleclient.org/projects/sipsimpleclient/repository/entry/patches/pjsip-webrtc_aec.patch) >> shows. PJMedia already has an interface for noise supression and AEC >> so I guess it was "easy" to implement there. We would implement it as >> a media bug or on the portaudio module itself? >> >> Also, if anyone knows if this will bring good results, would be >> great! AEC that just works is something that is really wanted on a >> softphone... >> >> >> Jo?o Mesquita >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From marketing at cluecon.com Mon Apr 15 21:15:06 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 15 Apr 2013 10:15:06 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes Message-ID: Greetings! Happy tax day to those in the USA - we hope all is well with your business. Speaking of business, I thought I would relay the interesting newsabout Dish Network making a bid for Sprint. Many here in North America will be keeping a close eye on this one. Whether or not this is just a big messor an opportunity for Sprint to become a "real" competitor to AT&T and Verizon remains to be seen. Regardless of the outcome, most of us here are hoping for a healthier Sprint so that we can avoid another duopoly. On last week's conference callwe decided to have a preliminary discussion so that we can prepare for this week . Dave Kompel will be showing us how to build rapidly FreeSWITCH applications in MS Visual Studio 2012 and have those run under mod_managed. Be sure to consult this documentso that you can get all the prerequisites installed in time for our call on Wednesday. In other news I would like to let everyone know that I spoke with Kashif Kahn over at Vestec . We are gearing up for the automatic speech recognition application building contest. The winners will be announced at ClueCon 2013 . The official contest page will be posted on the ClueCon website shortly. Stay tuned for more information and be ready to start building your ASR applications! The ClueCon 2013 call for speakers recently went out and we've had a number of submissions already. We look forwarding to hearing more talk ideas, so please send those in right away. In the meantime ClueCon registrationis now open so be sure to get signed up, and don't forget to book your room at the Hyatt Chicago Magnificent Mile hotel for only $169 per night. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130415/7b761ca4/attachment.html From msc at freeswitch.org Wed Apr 17 20:23:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 17 Apr 2013 09:23:12 -0700 Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today Message-ID: Hello all! Today's community conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_17 Today we will be using MS Visual Studio 2012 to build rapidly FreeSWITCH applications. We look forward to hearing from Dave Kompel on this discussion. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130417/0f6fca5b/attachment-0001.html From msc at freeswitch.org Thu Apr 18 03:09:34 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 17 Apr 2013 16:09:34 -0700 Subject: [Freeswitch-dev] Vestec ASR example apps Message-ID: Hello all! Has anyone out there been using Vestec ASR? I need to gather some basic how-to stuff to create a Vestec section on our wiki. We are gearing up for the Vestec ASR app contest and I am hoping to lower the barrier to entry for entrants by having a really nice "get-you-started" wiki page. Please email me off-list if you have done anything with Vestec and are willing to share some examples. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130417/3fa3b247/attachment.html From krice at freeswitch.org Fri Apr 19 17:31:42 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 19 Apr 2013 08:31:42 -0500 Subject: [Freeswitch-dev] Jira Open Tickets and Patches (IMPORTANT INFORMATION) Message-ID: For the last several weeks Ray and myself have embarked on a weekly meeting to go thru the outstanding tickets on Jira and work toward resolution on them. We have asked for reporters to provide updated information on many of them and a good number of those reporters have responded. To those that have updated their tickets as requested Thank You! To those ignoring the the requests for more information, if you have not responded to requests for more information within 14 days, your ticket will likely be closed unless you respond. This policy will be enforced on a ticket by ticket basis. With the core team making the final call. At some point in the future we will likely automate this process. Now, that being said, the goal here is not to just reduce the number of open tickets, but to make sure we are focusing efforts where we can actually do something about it. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130419/66ae8ab0/attachment.html From richard.screene at netdev.co.uk Fri Apr 19 17:48:47 2013 From: richard.screene at netdev.co.uk (Richard Screene) Date: Fri, 19 Apr 2013 14:48:47 +0100 Subject: [Freeswitch-dev] FreeSWITCH for WebRTC Message-ID: <92D97385-C2B2-42FB-A037-7AEB445A8124@netdev.co.uk> Is there any news on when WebRTC support will be added to FreeSWITCH? It would be awesome if we could host a video conference using FreeSWITCH. Regards, Richard From krice at freeswitch.org Fri Apr 19 17:53:31 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 19 Apr 2013 08:53:31 -0500 Subject: [Freeswitch-dev] FreeSWITCH for WebRTC In-Reply-To: <92D97385-C2B2-42FB-A037-7AEB445A8124@netdev.co.uk> Message-ID: Theres no news as to when other then its coming at some point in the future... On 4/19/13 8:48 AM, "Richard Screene" wrote: > Is there any news on when WebRTC support will be added to FreeSWITCH? > It would be awesome if we could host a video conference using FreeSWITCH. > > Regards, > Richard > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From mike at jerris.com Fri Apr 19 19:18:20 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 19 Apr 2013 11:18:20 -0400 Subject: [Freeswitch-dev] FreeSWITCH for WebRTC In-Reply-To: References: Message-ID: <92A14EB1-36A6-4ECC-8D48-D74BD009AAAC@jerris.com> Isn't the saying "no news is good news" ? On Apr 19, 2013, at 9:53 AM, Ken Rice wrote: > Theres no news as to when other then its coming at some point in the > future... > > > On 4/19/13 8:48 AM, "Richard Screene" wrote: > >> Is there any news on when WebRTC support will be added to FreeSWITCH? >> It would be awesome if we could host a video conference using FreeSWITCH. >> >> Regards, >> Richard >> From anthony.minessale at gmail.com Fri Apr 19 19:36:33 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 19 Apr 2013 10:36:33 -0500 Subject: [Freeswitch-dev] FreeSWITCH for WebRTC In-Reply-To: <92A14EB1-36A6-4ECC-8D48-D74BD009AAAC@jerris.com> References: <92A14EB1-36A6-4ECC-8D48-D74BD009AAAC@jerris.com> Message-ID: You're thinking of the Great Space Coaster right? http://www.youtube.com/watch?v=mAwVIZDAUF0 They should have used this guy as the spokesman for GPL. But he's probably not free. On Fri, Apr 19, 2013 at 10:18 AM, Michael Jerris wrote: > Isn't the saying "no news is good news" ? > > On Apr 19, 2013, at 9:53 AM, Ken Rice wrote: > > > Theres no news as to when other then its coming at some point in the > > future... > > > > > > On 4/19/13 8:48 AM, "Richard Screene" > wrote: > > > >> Is there any news on when WebRTC support will be added to FreeSWITCH? > >> It would be awesome if we could host a video conference using > FreeSWITCH. > >> > >> Regards, > >> Richard > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130419/ae1af876/attachment.html From james at freedomnet.co.nz Fri Apr 19 19:37:50 2013 From: james at freedomnet.co.nz (james jones) Date: Fri, 19 Apr 2013 11:37:50 -0400 Subject: [Freeswitch-dev] FreeSWITCH for WebRTC In-Reply-To: <92A14EB1-36A6-4ECC-8D48-D74BD009AAAC@jerris.com> References: <92A14EB1-36A6-4ECC-8D48-D74BD009AAAC@jerris.com> Message-ID: I hope so....*stares at empty directory* On Fri, Apr 19, 2013 at 11:18 AM, Michael Jerris wrote: > Isn't the saying "no news is good news" ? > > On Apr 19, 2013, at 9:53 AM, Ken Rice wrote: > > > Theres no news as to when other then its coming at some point in the > > future... > > > > > > On 4/19/13 8:48 AM, "Richard Screene" > wrote: > > > >> Is there any news on when WebRTC support will be added to FreeSWITCH? > >> It would be awesome if we could host a video conference using > FreeSWITCH. > >> > >> Regards, > >> Richard > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130419/e2d93bfd/attachment-0001.html From msc at freeswitch.org Sat Apr 20 03:58:16 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 19 Apr 2013 16:58:16 -0700 Subject: [Freeswitch-dev] Hunting down bugs with git bisect Message-ID: Hello gang! I'm soliciting feedback on this topic: using git bisect to find "the commit that broke it." I am particularly interested in anyone who has automated the process so that it will build, test, pull, build, test, pull, etc. automatically without human intervention. If you've done anything like that please let us know. I'd like to see git bisect used more for people who are reporting bugs with phrases like, "It stopped working when I updated so I went back to the old git version and it works fine there." Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130419/c4cb97c6/attachment.html From marketing at cluecon.com Tue Apr 23 22:44:21 2013 From: marketing at cluecon.com (Michael Collins) Date: Tue, 23 Apr 2013 11:44:21 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes Message-ID: A belated welcome to the week of April 22! News and notes is a day late because I spent all day yesterday working on the new FreeSWITCH book . Good news - we have submitted all the content! It's now just a matter of getting all the edits and reviews done. We're a little more than halfway through that process. We'll update you as more news becomes available. Last week we had a conference call thatlasted quite a while. Dave Kompel showed us some tricks with using MS Visual Studio 2012 to build FreeSWITCH applications. That discussion dovetailed into a more general discussion about some of the other things Dave has been doing with FreeSWITCH under Windows. In the weeks and months ahead, Dave will be doing some more Windows-only discussions outside of the main weekly conference call. Keep an eye on the mailing listfor more announcements. Tomorrow we have scheduled an engineer from JeraSoftwho will be talking about their routing and billing solution that has been tightly integrated with FreeSWITCH 1.2. If you have FreeSWITCH in a production environment where billing and routing are core elements then this discussion will definitely be of interest. Have a great rest of the week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130423/f818d1f5/attachment.html From krice at freeswitch.org Wed Apr 24 03:56:44 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 23 Apr 2013 18:56:44 -0500 Subject: [Freeswitch-dev] Jira and the PasteBin... [Please READ ME] Message-ID: Hey Guys, Please keep the following things in mind. Bugs should be reported via Jira. Yes its ok to ask on the mailing list to see if it you are just doing something wrong, but please open a bug on Jira if its a bug. A segfault is always a bug. (Even if you are doing it wrong, FreeSWITCH should not segfault) The Pastebin is a useful tool for sharing information between people, but please do not reference pastebin URLs in Jira Tickets. Pastebin entries are not guaranteed to survive for any period of time, and they can and will disappear at some point. This means of you open a ticket and you put a backtrace in the pastebin, if the pastebin is cleared so is your backtrace. (if you use the fscore_pb script to help collect the data, there is a download link right on the pastebin that will let you download a .txt file for that entry which you can then attach to the ticket ) Please Attach to jira ticket all backtraces, sip traces and logs as plain text files ( .txt files ) Also, please do not paste the results into the notes or the description of the issue. This just means we have to scroll down 97 screens of stuff to get the last notes on a ticket. These simple things will help out the developers and help make the bugs go away faster! Thanks! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130423/fc5bf65e/attachment.html From vhernando at systemonenoc.com Wed Apr 24 14:52:53 2013 From: vhernando at systemonenoc.com (Vicente Hernando) Date: Wed, 24 Apr 2013 12:52:53 +0200 Subject: [Freeswitch-dev] mod_lua speed up Message-ID: <5177B985.3010103@systemonenoc.com> Hello, looking at src/mod/languages/mod_lua/mod_lua.cpp I see executing lua_init lua_uninit at the beginning and end of every function, then a lua script is loaded from a buffer, compiled and executed. Another approach could be loading some lua scripts at the beginning and compile them. Then create a pool of lua states. Now everytime lua_init is called, instead of initializing lua again, it would take a lua state from the pool if available, otherwise it would create a new one. lua_uninit would reintegrate a lua state into the pool again. At the end, when module is finished being used, the whole pool should be freed. Do you thing it is a worthwhile idea or a crazy idea? Any points? Thanks in advance, Vicente. From msc at freeswitch.org Wed Apr 24 19:50:48 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 24 Apr 2013 08:50:48 -0700 Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today Message-ID: Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_24 The folks from JeraSoft will be doing a presentation on their routing/billing solution that works with FreeSWITCH. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130424/6065e6ea/attachment.html From mike at jerris.com Wed Apr 24 20:21:11 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Apr 2013 12:21:11 -0400 Subject: [Freeswitch-dev] mod_lua speed up In-Reply-To: <5177B985.3010103@systemonenoc.com> References: <5177B985.3010103@systemonenoc.com> Message-ID: If you profile this, how long do those actually take? how does that compare to whats needed to clear out a lua state fresh so there is no bleed over to the next usage? On Apr 24, 2013, at 6:52 AM, Vicente Hernando wrote: > Hello, > > looking at src/mod/languages/mod_lua/mod_lua.cpp > > I see executing lua_init lua_uninit at the beginning and end of every > function, then a lua script is loaded from a buffer, compiled and executed. > > > Another approach could be loading some lua scripts at the beginning and > compile them. Then create a pool of lua states. > > Now everytime lua_init is called, instead of initializing lua again, it > would take a lua state from the pool if available, otherwise it would > create a new one. > > lua_uninit would reintegrate a lua state into the pool again. > > > At the end, when module is finished being used, the whole pool should be > freed. > > > Do you thing it is a worthwhile idea or a crazy idea? Any points? > > > Thanks in advance, > Vicente. > > From vhernando at systemonenoc.com Wed Apr 24 20:58:45 2013 From: vhernando at systemonenoc.com (Vicente Hernando) Date: Wed, 24 Apr 2013 18:58:45 +0200 Subject: [Freeswitch-dev] mod_lua speed up In-Reply-To: References: <5177B985.3010103@systemonenoc.com> Message-ID: <51780F45.80900@systemonenoc.com> Hi Michael, the goal for us is to compile only once the lua script ( in our case it performs connections to several redis instances ). I have still not tested it yet, only wanted to know if it was a crazy idea, or was it just plain impossible. When I get the profiling done I will post it here. Kind regards, Vicente. On 04/24/2013 06:21 PM, Michael Jerris wrote: > If you profile this, how long do those actually take? how does that compare to whats needed to clear out a lua state fresh so there is no bleed over to the next usage? > > On Apr 24, 2013, at 6:52 AM, Vicente Hernando wrote: > >> Hello, >> >> looking at src/mod/languages/mod_lua/mod_lua.cpp >> >> I see executing lua_init lua_uninit at the beginning and end of every >> function, then a lua script is loaded from a buffer, compiled and executed. >> >> >> Another approach could be loading some lua scripts at the beginning and >> compile them. Then create a pool of lua states. >> >> Now everytime lua_init is called, instead of initializing lua again, it >> would take a lua state from the pool if available, otherwise it would >> create a new one. >> >> lua_uninit would reintegrate a lua state into the pool again. >> >> >> At the end, when module is finished being used, the whole pool should be >> freed. >> >> >> Do you thing it is a worthwhile idea or a crazy idea? Any points? >> >> >> Thanks in advance, >> Vicente. >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > From anthony.minessale at gmail.com Wed Apr 24 21:17:25 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 24 Apr 2013 12:17:25 -0500 Subject: [Freeswitch-dev] mod_lua speed up In-Reply-To: <51780F45.80900@systemonenoc.com> References: <5177B985.3010103@systemonenoc.com> <51780F45.80900@systemonenoc.com> Message-ID: Its not crazy but its lofty and prone to regressions. It would be similar to database pooling code where it would need to create more if you used up the cached ones etc. Lua has its own byte compiled format doesn't it? Usually when you get past the point that lua is not satisfying, it means you should code it in C. On Wed, Apr 24, 2013 at 11:58 AM, Vicente Hernando < vhernando at systemonenoc.com> wrote: > Hi Michael, > > the goal for us is to compile only once the lua script ( in our case it > performs connections to several redis instances ). > > I have still not tested it yet, only wanted to know if it was a crazy > idea, or was it just plain impossible. > > When I get the profiling done I will post it here. > > > Kind regards, > Vicente. > > On 04/24/2013 06:21 PM, Michael Jerris wrote: > > If you profile this, how long do those actually take? how does that > compare to whats needed to clear out a lua state fresh so there is no bleed > over to the next usage? > > > > On Apr 24, 2013, at 6:52 AM, Vicente Hernando < > vhernando at systemonenoc.com> wrote: > > > >> Hello, > >> > >> looking at src/mod/languages/mod_lua/mod_lua.cpp > >> > >> I see executing lua_init lua_uninit at the beginning and end of every > >> function, then a lua script is loaded from a buffer, compiled and > executed. > >> > >> > >> Another approach could be loading some lua scripts at the beginning and > >> compile them. Then create a pool of lua states. > >> > >> Now everytime lua_init is called, instead of initializing lua again, it > >> would take a lua state from the pool if available, otherwise it would > >> create a new one. > >> > >> lua_uninit would reintegrate a lua state into the pool again. > >> > >> > >> At the end, when module is finished being used, the whole pool should be > >> freed. > >> > >> > >> Do you thing it is a worthwhile idea or a crazy idea? Any points? > >> > >> > >> Thanks in advance, > >> Vicente. > >> > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130424/8bb78dd3/attachment.html From mike at jerris.com Wed Apr 24 21:27:53 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 24 Apr 2013 13:27:53 -0400 Subject: [Freeswitch-dev] mod_lua speed up In-Reply-To: <51780F45.80900@systemonenoc.com> References: <5177B985.3010103@systemonenoc.com> <51780F45.80900@systemonenoc.com> Message-ID: if it offers a 1ms improvement per call, then its crazy. If it offers a 10second improvement per call, its not. On Apr 24, 2013, at 12:58 PM, Vicente Hernando wrote: > Hi Michael, > > the goal for us is to compile only once the lua script ( in our case it > performs connections to several redis instances ). > > I have still not tested it yet, only wanted to know if it was a crazy > idea, or was it just plain impossible. > > When I get the profiling done I will post it here. > > > Kind regards, > Vicente. > > On 04/24/2013 06:21 PM, Michael Jerris wrote: >> If you profile this, how long do those actually take? how does that compare to whats needed to clear out a lua state fresh so there is no bleed over to the next usage? >> >> On Apr 24, 2013, at 6:52 AM, Vicente Hernando wrote: >> >>> Hello, >>> >>> looking at src/mod/languages/mod_lua/mod_lua.cpp >>> >>> I see executing lua_init lua_uninit at the beginning and end of every >>> function, then a lua script is loaded from a buffer, compiled and executed. >>> >>> >>> Another approach could be loading some lua scripts at the beginning and >>> compile them. Then create a pool of lua states. >>> >>> Now everytime lua_init is called, instead of initializing lua again, it >>> would take a lua state from the pool if available, otherwise it would >>> create a new one. >>> >>> lua_uninit would reintegrate a lua state into the pool again. >>> >>> >>> At the end, when module is finished being used, the whole pool should be >>> freed. >>> >>> >>> Do you thing it is a worthwhile idea or a crazy idea? Any points? >>> >>> >>> Thanks in advance, >>> Vicente. >>> >>> From marketing at cluecon.com Mon Apr 29 21:32:52 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 29 Apr 2013 10:32:52 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes Message-ID: Hello Friends, We are looking forward to a busy but good week of activity. Last week was busy as well. On last week's conference callwe had an engineer from JiraSoft join us to answer some questions about the newly released VCS 2.4.3. We are happy to have this new version as part of the FreeSWITCH ecosystem. This week we welcome Omar from OrecX . OrecX creates commercial and open source telephony recording solutions. We look forward to learning more about how we can use OrecX with FreeSWITCH. One other bit of good news: All of the content for the new FreeSWITCH bookhas been submitted to the publishers. We are doing a bit of editing on one of the chapters but otherwise we are basically done. We anticipate a June release. Take care and have another fantastic week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20130429/8eb7c17b/attachment.html