November 2013 - freeswitch-dev@lists.freeswitch.org

Re: [Freeswitch-dev] Freeswitch packaging for Raspberry Pi?
by Hans Bakker 15 Dec '13

15 Dec '13

Hi all, I'm searching for precompiled Freeswitch binaries for the Raspberry Pi. I found http://wiki.freeswitch.org/wiki/Debian_packages_buildscript and I've seen some attempts in this thread http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-on-Raspberry-Pi-Op…. Also I found some blogs explaining how to compile Freeswitch from source (manually or using the Fusionpbx script). However it would be much more convenient to have packaged binaries, to be able to install module packages instead of having to build them again, and to be able to quickly update the software. Could you please tell me whether there are any plans of automating Freeswitch packaging for the Raspberry Pi? Kind regards, Hans 2013/11/7 Hans Bakker <hansmbakker(a)gmail.com> > Hi all, > > I'm searching for precompiled Freeswitch binaries for the Raspberry Pi. > I've seen some attempts in this thread > http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-on-Raspberry-Pi-Op…. > Also I found some blogs explaining how to compile Freeswitch from source > (manually or using the Fusionpbx script). > > However it would be much more convenient to have packaged binaries. > > Could you please tell me whether there are any plans of packaging > Freeswitch for the Raspberry Pi? > > Kind regards, > > Hans >

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[Freeswitch-dev] Accessing Custom Header in REGISTER packets
by Mark Tsai 04 Dec '13

04 Dec '13

Hi All, In my REGISTER packets that's sent towards a FreeSwitch server, I am adding a custom header "X-CustomHeaderName: auth_token". The goal for my project is to access that token and conduct a verification through another API before allowing the REGISTER to pass. What's the best way to accomplish this? thanks, Miz

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[Freeswitch-dev] FreeSWITCH erroneous RTP timestamps in front of sofsip_cli client?
by Jean-Paul Iribarren 04 Dec '13

04 Dec '13

Hi FS wizards, first message on this list... Apologies if this post belongs to another section (please tell me if this is the case). First, a big thank you to all members of the team for this amazing piece of software. Great job here. Now, on for my issue: I am currently testing a Linux sofsip_cli SIP client on an embedded device, connected to a Win32 FreeSWITCH version, and I can't get correct RX audio on my platform. I ran the same test with a linphonec client on the same embedded device for comparison purposes, and the rx audio is good. Both clients end up using the same RTP configuration (PCMU/8000). Preliminary conclusion: my sofsip_cli client is buggy. But there is more... I ended up checking data exchanged between endpoints using Wireshark, and I was surprised to see that in front of sofsip_cli, FreeSWITCH seems to timestamp TX RTP packets incorrectly: it sends 160 samples/RTP packet, but the timestamps of consecutive TX packets are increased in 320 units steps, and FS sends 25 RTP packets per second on average. Consecutive RTP packets actually contain consecutive PCMU data (from the audio point of view), resulting in a sloooow, chopped rx audio coming out of the loudspeaker of my device. Using Wireshark embedded player to play the VoIP audio results in the same slow / chopped audio, which seems to prove that FS actually sent bad RTP. On the other hand, when I use linphonec in the very same conditions, I can see that FreeSWITCH sends 50 RTP packets per second on average. Each RTP packet contains 160 samples, and timestamps of consecutive are increased in 160 units steps, resulting in correct RX audio coming out of the loudspeaker of my device. FS logs in console window are the same for both cases, below is what I get for the sofsip_cli case: 2013-11-21 09:35:00.549253 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1009(a)10.0.0.57 [527f855e-3856-4805-b72c-da59b3a6d292] 2013-11-21 09:35:04.502530 [INFO] mod_dialplan_xml.c:558 Processing 1009 <1009>->1001 in context default 2013-11-21 09:35:04.533781 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *1 execute_extension::dx XML features 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *2 record_session::C:/Program Files/FreeSWITCH/recordings/1009.2013-11-21-09-35-04.wav 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *3 execute_extension::cf XML features 2013-11-21 09:35:04.549407 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *4 execute_extension::att_xfer XML features 2013-11-21 09:35:04.565032 [NOTICE] switch_ivr_originate.c:2699 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2013-11-21 09:35:04.565032 [NOTICE] switch_ivr_originate.c:2699 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2013-11-21 09:35:04.580658 [INFO] mod_dptools.c:3201 Originate Failed. Cause: USER_NOT_REGISTERED 2013-11-21 09:35:04.580658 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1009(a)10.0.0.57! 2013-11-21 09:35:04.580658 [NOTICE] mod_dptools.c:1225 Channel [sofia/internal/1009(a)10.0.0.57] has been answered 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1055 New Channel loopback/app=voicemail:default 10.0.0.57 1001-a [f2aa95f1-0631-4590-9fa4-c4dc1c96c88e] 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1053 Rename Channel loopback/app=voicemail:default 10.0.0.57 1001-a->loopback/voicemail-a [f2aa95f1-0631-4590-9fa4-c4dc1c96c88e] 2013-11-21 09:35:05.580696 [NOTICE] switch_channel.c:1055 New Channel loopback/voicemail-b [fc478cfd-e932-450a-9f50-2a5a76acc907] 2013-11-21 09:35:05.611948 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a! 2013-11-21 09:35:05.611948 [NOTICE] mod_dptools.c:1260 Pre-Answer loopback/voicemail-b! 2013-11-21 09:35:11.549676 [NOTICE] sofia.c:715 Hangup sofia/internal/1009(a)10.0.0.57 [CS_EXECUTE] [NORMAL_CLEARING] 2013-11-21 09:35:11.565301 [NOTICE] switch_ivr_bridge.c:733 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2013-11-21 09:35:11.565301 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2013-11-21 09:35:11.580927 [NOTICE] switch_core_session.c:1598 Session 5 (loopback/voicemail-a) Ended 2013-11-21 09:35:11.580927 [NOTICE] switch_core_session.c:1602 Close Channel loopback/voicemail-a [CS_DESTROY] 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1598 Session 4 (sofia/internal/1009(a)10.0.0.57) Ended 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close Channel sofia/internal/1009(a)10.0.0.57 [CS_DESTROY] 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1598 Session 6 (loopback/voicemail-b) Ended 2013-11-21 09:35:11.612178 [NOTICE] switch_core_session.c:1602 Close Channel loopback/voicemail-b [CS_DESTROY] I ran my tests by calling an unregistered extension (1001), therefore resulting in FS sending voice messaging audio, and hanging up on client after a few seconds. I have compared the SIP handshakes for both clients, and I can't see any significant difference (except that linphonec supports several codecs, while sofsip_cli only supports G.711/PCMU). Regarding FS config (this may or may not be relevant), I had to tweak it a bit to support sofsip_cli. I added the two following lines in the dialplan for the relevant extensions (file default.xml): <action application="set" data="disable_rtp_auto_adjust=true"/> <action application="set" data="zrtp_secure_media=false"/> I also changed the two following lines to add video codecs in file vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,H263,H263-1998,H264"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,H263,H263-1998,H264"/> The rest of the config is plain vanilla. The (small) Wireshark traces of both tests can be downloaded from Box: - linphonec trace (FS TX RTP ok): file linphone.dac.pcap is available at https://app.box.com/s/63p8ayrr3dc6w1d9wm33 - sofsip_cli trace (FS TX RTP ko): file ssc.pcap is available at https://app.box.com/s/z1cde37xg5teh9d690oy My device has IP address 10.0.0.186, and FS is installed on my PC (running Win XP Pro) at 10.0.0.57. My FreeSWITCH version is "1.5.6b+git~20131101T202135Z~2589bf7750~32bit (git 2589bf7 2013-11-01 20:21:35Z 32bit)" Sorry for the long post (I tried to give all relevant info), and thank you in advance for your comments and help on this weird issue. -- JPI

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[Freeswitch-dev] Connect to SIP Trunking
by Roamer Roamer 02 Dec '13

02 Dec '13

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

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[Freeswitch-dev] PostgreSQL for windows build
by Ali R. 27 Nov '13

27 Nov '13

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

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[Freeswitch-dev] Error handling / Error callbacks in Freeswitch
by Bhavesh 26 Nov '13

26 Nov '13

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

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[Freeswitch-dev] Is there any error handler in freeswitch
by Bhavesh 25 Nov '13

25 Nov '13

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

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[Freeswitch-dev] Hangup hook gets destroyed when calling a script in conference caller controls
by Melanie Treitinger 18 Nov '13

18 Nov '13

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

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[Freeswitch-dev] The gentones application generates a very bad quality sound
by Anton 17 Nov '13

17 Nov '13

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

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[Freeswitch-dev] FreeSWITCH 1.2.15 Released!!!
by Ken Rice 16 Nov '13

16 Nov '13

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

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