August 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Strange media behavior on WebRTC
by Oleg Stolyar 05 Aug '14

05 Aug '14

Hi guys, I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that: 1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference. 2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears. This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both. I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library. Has anyone run into this? Any advice? Here is the relevant excerpt from my dialplan. <condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>

4 49

Limit Database
by Andre 05 Aug '14

05 Aug '14

HI, on Windows I'm trying to use Limit with Microsoft SQL DB but it's still using the sqlight database. I have mod_Db and Mod_Hash loaded and under db.config.xml I have my odbc setup like this <param name="odbc-dsn" value="sqldb64::" /> I know the odbc works because I have the core database setup the same <param name="odbc-dsn" value="sqldb64::" /> <param name="core-dbtype" value="MSSQL"/> Is there a step that I'm missing? Thanks

11 20

esl_recv_event() from two threads on the same handle
by Shlomi Agiv 05 Aug '14

05 Aug '14

I'm using the C ESL events api. I have one threads that continuously checks for updates using esl_recv_event() I have another thread that sometimes needs to inject commands, that uses esl_send_recv() on the same handle. It looks like on occasion, both get stuck and stop receiving events. Is it possible for two threads to access the handle? maybe I should open two connections to freeswitch esl? or use esl_send_recv_timed()? stack trace: #0 0xb6f3f974 in __lll_lock_wait () from /lib/arm-linux-gnueabi/libpthread.so.0 #1 0xb6f39858 in pthread_mutex_lock () from /lib/arm-linux-gnueabi/libpthread.so.0 #2 0x000f2f2c in esl_mutex_lock (mutex=<optimized out>) at src/esl_threadmutex.c:200 #3 0x000efb70 in esl_recv_event (handle=0x11b67c, check_q=0, save_event=0xb6c47da4) at src/esl.c:1262 #4 0x0001ebbc in EslHandler::initThread (this=0x11b670) at eslHandler.cpp:95 #5 0x0001ec88 in EslHandler::entryPoint (eslHandler=0x11b670) at eslHandler.cpp:110 #6 0xb6f36b04 in start_thread () from /lib/arm-linux-gnueabi/libpthread.so.0 #7 0xb6d25a2c in ?? () from /lib/arm-linux-gnueabi/libc.so.6 #8 0xb6d25a2c in ?? () from /lib/arm-linux-gnueabi/libc.so.6 Backtrace stopped: previous frame identical to this frame (corrupt stack?) (gdb) thread 1 [Switching to thread 1 (Thread 0xb6c4a000 (LWP 3020))] #0 0xb6d17c04 in poll () from /lib/arm-linux-gnueabi/libc.so.6 (gdb) info stack #0 0xb6d17c04 in poll () from /lib/arm-linux-gnueabi/libc.so.6 #1 0x000ef92c in esl_wait_sock (sock=<optimized out>, ms=<optimized out>, flags=<optimized out>) at src/esl.c:929 #2 0x000ef988 in handle_recv (handle=0x11b67c, data=0x11b91c, datalen=65535) at src/esl.c:1230 #3 0x000efc84 in esl_recv_event (handle=0x11b67c, check_q=<optimized out>, save_event=0x12c120) at src/esl.c:1323 #4 0x000f0ba4 in esl_send_recv_timed (ms=0, cmd=<optimized out>, handle=0x11b67c) at src/esl.c:1554 #5 esl_send_recv_timed (handle=0x11b67c, cmd=<optimized out>, ms=0) at src/esl.c:1521 #6 0x0001f2e0 in EslHandler::sendAtCommand (this=0x11b670, atCmd=...) at eslHandler.cpp:204 #7 0x0001f86c in EslHandler::getImsiFromSim (this=0x11b670) at eslHandler.cpp:262 #8 0x0001450c in FsControl::checkMessages (this=0x11b2a0) at fsControl.cpp:435 #9 0x000124cc in FsControl::handleEslCustomEvent (this=0x11b2a0, mapEventReceived=...) at fsControl.cpp:376 #10 0x0000f334 in FsControl::handleEslEvent (this=0x11b2a0, mapEventReceived=...) at fsControl.cpp:226 #11 0x0000e7a8 in FsControl::OnWaitHandle (this=0x11b2a0, whp=0x13aea8, data=...) at fsControl.cpp:140 #12 0x000d230c in os::Schedualed::OnEvent (this=0x11b2a0, event=0x13aea8, data=...) at cpp/os.cpp:1578 #13 0x000cd1d0 in os::Schedualer::mainloop (this=0x11b348, exitOnNoneManaged=false) at cpp/os.cpp:446 #14 0x000161e0 in main (argc=3, argv=0xbea16904) at fsControl.cpp:603

3 3

Re: [Freeswitch-users] How to get real-time rtp audio and video ports
by takashi higashi 04 Aug '14

04 Aug '14

Hi, for example, When sip://1000@192.168.100.1 user and sip://1001@192.168.100.1 is talking, I want to get 1000 and 1001 user’s audio and video rtp packet using ffmpeg. So I have to specify 1000,1001 users rtp ports, right? my goal is that encoding from rtp to m-jpeg and transfer to node.js. How can I get 1000,1001 user’s audio and video rtp ports? Thanks, > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: How to get real-time rtp audio and video ports > (Michael Jerris) > 2. Re: Latest master and latest v1.4 consume more CPU for > conferences (Michael Jerris) > 3. Re: Latest master and latest v1.4 consume more CPU for > conferences (Oleg Stolyar) > > 差出人: Michael Jerris <mike(a)jerris.com> > 件名: Re: [Freeswitch-users] How to get real-time rtp audio and video ports > 日付: 2014年8月1日 1:29:36 JST > 宛先: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 返信先: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > I don't understand your question. That is the port for the rtp stream. What do you mean what is it? > > > On Jul 30, 2014, at 9:34 PM, takashi higashi <higashi(a)itsherpa.com> wrote: > >> Hey guys, >> I sent e-mail to freeswitch developer about how to get rtp packet with ffmpeg. and I asked to him below question. >> How can I get rtp packet with ffmpeg? I want to trancoding real time rtp packet to motion jpeg and transfer node.js. >> and then, he answered like this, >> The following works for me: >> ffmpeg -re -f lavfi -i aevalsrc="sin(400*2*PI*t)" -ar 8000 -f mulaw -f rtp rtp://127.0.0.1:6666 >> ffmpeg -i pcmu.sdp -acodec copy /tmp/a.wav >> or >> ffmpeg -i bingbing.mp4 -an -vcodec h264 -filter:v scale=1280:720 -f rtp rtp://127.0.0.1:6668 >> looks like pcmu is not a valid codec for mp4, so I use aac >> ffmpeg -i pcmu-264.sdp -acodec libfaac -vcodec copy /tmp/a.mp4 >> >> So I asked about what is 6666 ports? How can I get rtp real-time audio and video ports? >> but I couldn't get answer... >> Somebody help me, how to get real-time rtp audio and video ports? please. > > > > > > > 差出人: Michael Jerris <mike(a)jerris.com> > 件名: Re: [Freeswitch-users] Latest master and latest v1.4 consume more CPU for conferences > 日付: 2014年8月1日 1:54:34 JST > 宛先: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 返信先: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > There is no reason at all to think it has anything to do with the crash. For the crash, simply get a backtrace of it so we can take a look at what is going on with the crash. > > On Jul 31, 2014, at 11:54 AM, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: > >> Well, it gets a little complicated. I need to enable multiple simultaneous eavesdroppers and the ability to include one extra person into the conversation. Let's call it Leg C. Then Leg B should be able to drop off leaving Leg A and Leg B talking. Additionally I need the ability for Leg B and Leg C to talk to each other putting Leg A on hold. >> >> I considered using the eavesdrop tool for this which would give me the ability for the eavesdropper to talk to Leg A, Leg B or both. However, I was not sure that having Leg A drop off would work and ultimately decided that the simplicity of implementing this with conferences is worth the CPU hit. >> >> Again, the only reason I am concerned with the current additional CPU jump is because it's unexplained and I suspect it has something to do with the crash I experienced. >> >> >> On Thu, Jul 31, 2014 at 8:35 AM, Steven Ayre <steveayre(a)gmail.com> wrote: >> Just one extra? >> >> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_three_way >> >> >> On 31 July 2014 15:04, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: >> That's a whole different discussion. Mostly because I need the ability to easily add another person to the call. >> >> >> On Thu, Jul 31, 2014 at 6:48 AM, Russell Treleaven <rtreleaven(a)bunnykick.ca> wrote: >> Why do you have two person conferences instead of bridged calls? >> >> Sent from my BlackBerry® PlayBook™ >> www.blackberry.com >> >> From: "Oleg Stolyar" <olegstolyar(a)gmail.com> >> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> >> Sent: 31 July, 2014 9:40 AM >> Subject: Re: [Freeswitch-users] Latest master and latest v1.4 consume more CPU for conferences >> >> I am not sure we are on the same page with the numbers. The relative increase is about 30%-40% >> >> The 0.3% is the absolute increase per conference. CPU usage per single two-member conference went from 1.2%-1.3% to 1.6%-2.0% >> >> I am using AWS m3.xlarge instances with CentOS. It's a 4-core system, so the increase in CPU of usage of 30% would reduce the number of calls each instance can take. The increase itself would not necessarily be a huge deal except if it's unexplained, it may be an indicator that something is wrong. As I said in my original email, the system running the recent code also crashed less than 24 hours after starting to take live calls. I wanted to call out the increase as it seems likely to me (although by no means certain) that the two are related. >> >> The previous FreeSWITCH version I used have always been rock solid - this was the first time FreeSWITCH crashed on me. >> >> >> On Thu, Jul 31, 2014 at 6:21 AM, Brian West <brian(a)freeswitch.org> wrote: >> What kind of system are you running on where 0.2 to 0.3% CPU increase would really matter? >> >> >> On Wed, Jul 30, 2014 at 9:39 PM, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: >> Of course, and it did when I first made the rate change. It went from 1% CPU to the 1.2%-1.3% in the master from May. The problem I am running into now is that there is another jump in CPU with no additional config change. >> >> If nothing comes to mind, I will patch the May master with the fix for WebRTC password size to fix my immediate issue and then try to find the time to bisect git and pinpoint what happened. >> >> >> On Wed, Jul 30, 2014 at 7:26 PM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: >> It is very likely to add to the cpu to transcode everything to 12000hz >> >> >> >> On Wed, Jul 30, 2014 at 6:12 PM, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: >> Hi Anthony, thank you for responding! >> >> I apologize for the previous email - I retracted it too soon - the CPU simply jumped down for a bit then went back up. >> >> Here are the full versions for both the master where it happened and v1.4. Is this enough? >> >> FreeSWITCH Version 1.5.14b+git~20140729T132247Z~af64da86f2~64bit (git af64da8 2014-07-29 13:22:47Z 64bit) >> FreeSWITCH Version 1.4.7+git~20140724T230506Z~9d1c0f7f3d~64bit (git 9d1c0f7 2014-07-24 23:05:06Z 64bit) >> >> It does not occur on this master: >> FreeSWITCH Version 1.5.13b+git~20140522T175648Z~9d99c34682~64bit (git 9d99c34 2014-05-22 17:56:48Z 64bit) >> >> I figured I would try this email before bisecting git but it will likely be my next step. >> >> On the subject of the 12000 rate related bug - I'd love to discuss this more with you - perhaps I'll try to get a little bit of your time at ClueCon to just show you the problem. It actually happens on every platform I tested it on except Debian 7. Just in the last couple of days I reproduced it on Ubuntu 12.04 and Ubuntu 14.04. I only mentioned it in this thread in case the 12000 sample rate conference on 8000 codecs is relevant to the CPU increase. >> >> >> >> On Wed, Jul 30, 2014 at 3:46 PM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: >> I'm not really sure how to pinpoint .8% cpu increase. You would maybe have to git bisect? I believe you already said this in another email then retracted it. So with all the back and forth it starts to sound like you have misinformation. >> >> You can't use vague references like "latest" which changes every 10 min. >> You need to try the most current master (TODAY then recite the hash) with a clean build and have a valid backtrace. >> The things you are saying are inconsistent so I think you need to rule out build skew from jumping around so many versions. >> Your 12000 thing also makes no sense at all and does not occur for anyone else which suggests you have environmental problem related to the way your system is setup. >> >> >> >> >> On Wed, Jul 30, 2014 at 5:31 PM, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: >> Hi guys, >> >> I have my FreeSWITCH running a bunch of conferences with two or three users using PCMU/8000 from all the user agents. I have the conference sample rate set to 12000 to get around a strange bug I reported a couple of months ago. >> >> In my old installation with master from May 22 2014, the servers consumed about 1.2%-1.3% CPU per two-person conference. >> >> In the latest master AND in the latest v1.4 it went up to about 1.6%-2.0% CPU per two-person conference. CPU usage by the FreeSWITCH process also jumps up and down more. >> >> Additionally (not sure if it's related) the latest master crashed less than 24 hours after it started taking live traffic. I ma trying to get the core dump and see if v1.4 also crashes but including the info here in case it gives someone an idea of what's going on with the CPU. >> >> Any ideas as to what's causing the increased CPU usage would be greatly appreciated. >> >> ___ > > > > 差出人: Oleg Stolyar <olegstolyar(a)gmail.com> > 件名: Re: [Freeswitch-users] Latest master and latest v1.4 consume more CPU for conferences > 日付: 2014年8月1日 2:06:47 JST > 宛先: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 返信先: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Yes, I enabled core dump and will get the backtrace if it crashes again. I need to make preparations for the rest of my system to handle FreeSWITCH crashes better before I put the new code back in production, so it may take a while. > > If I find the time I will also try to bisect git to find the commit that caused this as Anthony suggested. > > > On Thu, Jul 31, 2014 at 9:54 AM, Michael Jerris <mike(a)jerris.com> wrote: > There is no reason at all to think it has anything to do with the crash. For the crash, simply get a backtrace of it so we can take a look at what is going on with the crash. > > On Jul 31, 2014, at 11:54 AM, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: > >> Well, it gets a little complicated. I need to enable multiple simultaneous eavesdroppers and the ability to include one extra person into the conversation. Let's call it Leg C. Then Leg B should be able to drop off leaving Leg A and Leg B talking. Additionally I need the ability for Leg B and Leg C to talk to each other putting Leg A on hold. >> >> I considered using the eavesdrop tool for this which would give me the ability for the eavesdropper to talk to Leg A, Leg B or both. However, I was not sure that having Leg A drop off would work and ultimately decided that the simplicity of implementing this with conferences is worth the CPU hit. >> >> Again, the only reason I am concerned with the current additional CPU jump is because it's unexplained and I suspect it has something to do with the crash I experienced. >> >> >> On Thu, Jul 31, 2014 at 8:35 AM, Steven Ayre <steveayre(a)gmail.com> wrote: >> Just one extra? >> >> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_three_way >> >> >> On 31 July 2014 15:04, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: >> That's a whole different discussion. Mostly because I need the ability to easily add another person to the call. >> >> >> On Thu, Jul 31, 2014 at 6:48 AM, Russell Treleaven <rtreleaven(a)bunnykick.ca> wrote: >> Why do you have two person conferences instead of bridged calls? >> >> Sent from my BlackBerry® PlayBook™ >> www.blackberry.com >> >> From: "Oleg Stolyar" <olegstolyar(a)gmail.com> >> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> >> Sent: 31 July, 2014 9:40 AM >> Subject: Re: [Freeswitch-users] Latest master and latest v1.4 consume more CPU for conferences >> >> I am not sure we are on the same page with the numbers. The relative increase is about 30%-40% >> >> The 0.3% is the absolute increase per conference. CPU usage per single two-member conference went from 1.2%-1.3% to 1.6%-2.0% >> >> I am using AWS m3.xlarge instances with CentOS. It's a 4-core system, so the increase in CPU of usage of 30% would reduce the number of calls each instance can take. The increase itself would not necessarily be a huge deal except if it's unexplained, it may be an indicator that something is wrong. As I said in my original email, the system running the recent code also crashed less than 24 hours after starting to take live calls. I wanted to call out the increase as it seems likely to me (although by no means certain) that the two are related. >> >> The previous FreeSWITCH version I used have always been rock solid - this was the first time FreeSWITCH crashed on me. >> >> >> On Thu, Jul 31, 2014 at 6:21 AM, Brian West <brian(a)freeswitch.org> wrote: >> What kind of system are you running on where 0.2 to 0.3% CPU increase would really matter? >> >> >> On Wed, Jul 30, 2014 at 9:39 PM, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: >> Of course, and it did when I first made the rate change. It went from 1% CPU to the 1.2%-1.3% in the master from May. The problem I am running into now is that there is another jump in CPU with no additional config change. >> >> If nothing comes to mind, I will patch the May master with the fix for WebRTC password size to fix my immediate issue and then try to find the time to bisect git and pinpoint what happened. >> >> >> On Wed, Jul 30, 2014 at 7:26 PM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: >> It is very likely to add to the cpu to transcode everything to 12000hz >> >> >> >> On Wed, Jul 30, 2014 at 6:12 PM, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: >> Hi Anthony, thank you for responding! >> >> I apologize for the previous email - I retracted it too soon - the CPU simply jumped down for a bit then went back up. >> >> Here are the full versions for both the master where it happened and v1.4. Is this enough? >> >> FreeSWITCH Version 1.5.14b+git~20140729T132247Z~af64da86f2~64bit (git af64da8 2014-07-29 13:22:47Z 64bit) >> FreeSWITCH Version 1.4.7+git~20140724T230506Z~9d1c0f7f3d~64bit (git 9d1c0f7 2014-07-24 23:05:06Z 64bit) >> >> It does not occur on this master: >> FreeSWITCH Version 1.5.13b+git~20140522T175648Z~9d99c34682~64bit (git 9d99c34 2014-05-22 17:56:48Z 64bit) >> >> I figured I would try this email before bisecting git but it will likely be my next step. >> >> On the subject of the 12000 rate related bug - I'd love to discuss this more with you - perhaps I'll try to get a little bit of your time at ClueCon to just show you the problem. It actually happens on every platform I tested it on except Debian 7. Just in the last couple of days I reproduced it on Ubuntu 12.04 and Ubuntu 14.04. I only mentioned it in this thread in case the 12000 sample rate conference on 8000 codecs is relevant to the CPU increase. >> >> >> >> On Wed, Jul 30, 2014 at 3:46 PM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: >> I'm not really sure how to pinpoint .8% cpu increase. You would maybe have to git bisect? I believe you already said this in another email then retracted it. So with all the back and forth it starts to sound like you have misinformation. >> >> You can't use vague references like "latest" which changes every 10 min. >> You need to try the most current master (TODAY then recite the hash) with a clean build and have a valid backtrace. >> The things you are saying are inconsistent so I think you need to rule out build skew from jumping around so many versions. >> Your 12000 thing also makes no sense at all and does not occur for anyone else which suggests you have environmental problem related to the way your system is setup. >> >> >> >> >> On Wed, Jul 30, 2014 at 5:31 PM, Oleg Stolyar <olegstolyar(a)gmail.com> wrote: >> Hi guys, >> >> I have my FreeSWITCH running a bunch of conferences with two or three users using PCMU/8000 from all the user agents. I have the conference sample rate set to 12000 to get around a strange bug I reported a couple of months ago. >> >> In my old installation with master from May 22 2014, the servers consumed about 1.2%-1.3% CPU per two-person conference. >> >> In the latest master AND in the latest v1.4 it went up to about 1.6%-2.0% CPU per two-person conference. CPU usage by the FreeSWITCH process also jumps up and down more. >> >> Additionally (not sure if it's related) the latest master crashed less than 24 hours after it started taking live traffic. I ma trying to get the core dump and see if v1.4 also crashes but including the info here in case it gives someone an idea of what's going on with the CPU. >> >> Any ideas as to what's causing the increased CPU usage would be greatly appreciated. >> >> ___ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org いつもお世話になっております。アイティーシェルパ東です。 ============================================= 　株式会社アイティーシェルパ　　東　高志　〒810-0073 　福岡県福岡市中央区舞鶴２−２−１１　　　　　　　　　　富士ビル赤坂３Ｆ　TEL：092-737-5755　FAX：092-737-5765 　E-mail：higashi(a)itsherpa.com =============================================

2 1

compiling from git fails (502)
by Peter Villeneuve 04 Aug '14

04 Aug '14

Hi, I just tried a git clone with the new stash url and there seems to be a problem server side. Has the url changed again or is the server having issue? /usr/local/src# git clone https://stash.freeswitch.org/scm/fs/freeswitch.git Cloning into 'freeswitch'... error: The requested URL returned error: 502 while accessing https://stash.freeswitch.org/scm/fs/freeswitch.git/info/refs fatal: HTTP request failed Thanks

2 2

Transferring to a conference asks for PIN only on leg A
by Iskren Hadzhinedev 04 Aug '14

04 Aug '14

Hey everyone! I have the following scenario: Bridged two-legged call between A and B in progress. The dialplan has an extension for transferring to a conference with the following rule: <action application="bind_meta_app" data="1 a i transfer::-both 8001 XML mydomain"/> When A presses /*1/ both legs are getting transferred to 8001 which is a PIN protected conference. When this happens, leg B is already in the conference room and if it’s empty, music on hold starts. At the same time leg A is prompted for a password. Am I missing something obvious here? Here’s the conference profile: <profile name="myconference"> <param name="cdr-log-dir" value="auto"/> <param name="domain" value="mydomain"/> <param name="rate" value="8000"/> <param name="interval" value="20"/> <param name="energy-level" value="300"/> <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/> <param name="alone-sound" value="conference/conf-alone.wav"/> <param name="moh-sound" value="$${hold_music}"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="kicked-sound" value="conference/conf-kicked.wav"/> <param name="locked-sound" value="conference/conf-locked.wav"/> <param name="is-locked-sound" value="conference/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/> <param name="pin-sound" value="conference/conf-pin.wav"/> <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/> <param name="pin" value="12345"/> <param name="moderator-pin" value="54321"/> <param name="moderator-controls" value="admin"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="true"/> <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> </profile> FreeSWITCH Version 1.5.13b+git~20140621T003222Z~47891d5caa~64bit (git 47891d5 2014-06-21 00:32:22Z 64bit) Thank you! Kind regards, -- *Iskren Hadzhinedev*

1 0

Compiling mod_python on windows for Python 33
by Joseph L. Casale 04 Aug '14

04 Aug '14

Hi, I have a project where I would like to replace an asterisk implementation with one of freeswitch, this depends on compiling mod_python against 33 on windows. My knowledge around c is limited and my few attempts were not successful. Can anyone point me along? Thanks, jlc

1 0

rfc3822 dtmf in esl events
by Grant Bagdasarian 04 Aug '14

04 Aug '14

Hello, Does the ESL generate DTMF events when the DTMF is sent/received using RFC3822? I don't see any DTMF events when DTMF is received by FS from the caller. Any suggestions? Regards, Grant

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I'm come from Asterisk I need help to start
by matanya cohen 04 Aug '14

04 Aug '14

I install 1.15.12b Windows version, and I not find out how start, I have many question after I read books and articles: 1. The Windows ver is intended for Production, or only for test? It's tested on real system? 2. FS_cli not show enough lines on console. I not try connect by putty: 127.0.0.1 : 8021 and bot connect? 3. I try register X-lite with: user:1000 pass:222 domain:127.0.0.1 after I Edit C:\Program Files (x86)\FreeSWITCH\conf\directory\default\ 1000.xml to: <param name="password" value="222"/>, and reload it by F6. The error is: 401 unauthorized, I not see on log file any line about trying register? 4. I not see any forum on the internet to ask my problems Thanks

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Polycom hold problems?
by Yehavi Bourvine 04 Aug '14

04 Aug '14

Hi, After upgrading from Freeswitch 1.2.23 to 1.4.6 my users started to complain that they cannot use hold on incoming sessions on Polycom phones; when they try to return from hold the call is disconnected and I get a console error "CODEC renogotiation error". Our Polycom phones are set to offer SRTP, thus, on outgoing call they use SRTP but not on incoming. In the above scenario, when the user tries to return from hold, the phone sends SAVP with SRTP, although the original call was not encrypted, and this causes this error message. I had to disable SRTP on the Polycoms to make hold working again. BTW, this happens with Polycom 3.3.4 but not with 4.0.5 (but I cannot upgrade to it since it behaves erratically). Anyone else noticed this as well? Thanks, __Yehavi:

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