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FreeSwitch and BlueBox - cannot connect and no SIP Registration
by garcia16＠gmail.com 23 Nov '11

23 Nov '11

Hello, I have recently installed FreeSwitch with Blue.Box and I have two issues with them : 1. Module SIP Registration does not work, i mean when i click on him, nothing shows up. 2. When i run "sofia status" command on Freeswitch : Name Type Data State ================================================================================================= sipinterface_1 profile sip:mod_sofia@10.0.2.15:5090 RUNNING (0) sipinterface_3 profile sip:mod_sofia@10.0.2.15:5080 RUNNING (0) sipinterface_2 profile sip:mod_sofia@10.0.2.15:5070 RUNNING (0) ================================================================================================= My question is : how connect to my Freeswitch. I would also like to add that i am not able to add any gateway.

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Conference and Speex Wideband audio quality problem
by Richard Alam 23 Nov '11

23 Nov '11

Hi Everyone, Hopefully, the answer to my problem isn't in the book as I've yet to order it. :) First off, I'm one ot the devs of BigBlueButton (http://code.google.com/p/bigbluebutton/) We currently use Asterisk for voice conference. Leif Jackson made BBB work with FreeSWITCH. Now, from BBB, a user can join a voice conference in FS. We have a java SIP client (based on Red5Phone) that calls FS. All is working well for Ulaw. I am now trying to make Speex WB work since Flash Player (where our client runs) supports Speex WB. Using Speex WB frees the java SIP client from transcoding between Nellymoser and Ulaw. The java SIP client just takes the speex audio data from RTMP and sends it to FS through RTP. However, the sound quality is bad. It is choopy and robotic. Let me describe what I have done and what I found from debugging so far. I modified vars.xml to add the speex codec. ================ <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@16000h@20i,speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@16000h@20i,PCMU,PCMA,GSM"/> Setup the conference in dialplan/public.xml ================== <extension name="bbb_conferences"> <condition field="destination_number" expression="^(7\d{4})$"> <action application="answer"/> <action application="conference" data="$1@wideband"/> </condition> </extension> When I start up freeswitch, it looks like it's loading the speex codec fine. Full console log is here http://pastebin.com/rnHfgQYj =============== 2010-07-30 16:04:10.620375 [NOTICE] switch_loadable_module.c:251 Adding Application 'AvoidingDeadlock' 2010-07-30 16:04:10.620658 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_g723_1.so **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.620726 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_g729.so **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.621189 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_amr.so **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.621547 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_ilbc] 2010-07-30 16:04:10.621618 [NOTICE] switch_loadable_module.c:185 Adding Codec 'iLBC' (iLBC) 8000hz 30ms 2010-07-30 16:04:10.621647 [NOTICE] switch_loadable_module.c:185 Adding Codec 'iLBC' (iLBC) 8000hz 20ms 2010-07-30 16:04:10.622043 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_speex] 2010-07-30 16:04:10.622108 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 32000hz 20ms 2010-07-30 16:04:10.622135 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 16000hz 20ms 2010-07-30 16:04:10.622160 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 8000hz 20ms 2010-07-30 16:04:10.622513 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_h26x] When I call into the conference, it looks like mod_conference does not use speex codec but L16@16000hz. ====== 2010-07-30 16:05:56.874138 [DEBUG] sofia.c:4317 Channel sofia/external/75115(a)192.168.0.166 entering state [completed][200] 2010-07-30 16:05:56.874138 [DEBUG] switch_core_session.c:647 Send signal sofia/external/75115(a)192.168.0.166 [BREAK] 2010-07-30 16:05:56.874138 [DEBUG] switch_channel.c:2494 (sofia/external/75115(a)192.168.0.166) Callstate Change RINGING -> ACTIVE 2010-07-30 16:05:56.874138 [NOTICE] mod_dptools.c:746 Channel [sofia/external/75115(a)192.168.0.166] has been answered EXECUTE sofia/external/75115(a)192.168.0.166 conference(75115@wideband) 2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5237 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5282 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2010-07-30 16:05:56.877940 [DEBUG] switch_core_codec.c:116 sofia/external/75115(a)192.168.0.166 Push codec L16:10 tport_wakeup_pri(0x95bfd98): events IN Is this the correct behavior? I've tried using xlite to connect to the conference, and the audio is good. Also, when connecting from our client, the audio file that gets played by FS to notify how many users are in the conference is good. It's my voice that's bad. I tried to use the echo application instead of the conference. When I call from the BBB client, the sound quality is good. I appreciate any tips on how I can fix the problem or proceed to figure out how to fix it. Thank you very much. Richard -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton

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Wednesday Conf Call - SIP Presence?
by Michael Collins 23 Nov '11

23 Nov '11

Hi all, I would like to have someone speak on the subject of SIP presence this coming Wednesday. If you are up for giving the community a brief introduction to the concepts involved with SIP presence please let me know. Also, if you are experienced and/or knowledgeable on the subject but would rather not give a presentation also let me know as you may be in a position to type up some material that could be used by the presenter. Please email me off list. Thanks all! -MC

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Kernel timer ...those with experience please ?
by David Wafula 23 Nov '11

23 Nov '11

Hi all, I have very recent git (20th Nov 2011) on linode instance on Ubuntu 10.04, kernel 2.6.39. When alone in a conference, the music plays very neatly and clear. When another person joins the conference, then everything becomes distorted, cant even hold a simple conversion. Reading around, everyone attributes this to kernel timer. Could the kind experienced souls share their knowledge on kernel timer, how it affects freeswitch, and how best to correct it please. BTW, am considering during new image build with the latest kernel, will it help ? David Wafula

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P-Asserted-Identity: "Outbound Call" on groupcall
by Erik Dekkers 23 Nov '11

23 Nov '11

Hi List, Following my recent message on this list "P-Asserted-Identity: "Outbound Call" on internal calls" I've managed to get the phones to display the correct callee name. Unfortunately adding origination_callee_id_name to the groupcall doesn't work since I can't predict which groupmember will pickup that call. Is there a way to make this work correctly? Regards, Erik Dekkers (wvds-nl)

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When SSL/TLS enabled on FreeSwitch - sofia profile won't load
by Phil Quesinberry 23 Nov '11

23 Nov '11

I'm relatively new to FreeSwitch and am seriously impressed. That said, I'm having trouble getting TLS/SSL to work. When I enable it for a particular profile, that profile won't load. Running CentOS 5.7 - ModSSL, OpenSSL, and OpenSSL-devel are installed and the latest version. I did a bootstrap/configure/make after verifying that the needed SSL packages were installed but with the same result. Trying to restart the profile with TLS enabled from the FS CLI gives: Invalid Profile [internal]. I should mention that I'm running the FusionPBX front end to make it more convenient to perform the simpler admin tasks, just in case it matters. I did a make current late Sunday night. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f2cf68b 2011-11-20 18-40-41 -0500 Any suggestions as to what the problem might be or ways to go about tracking it down would be greatly appreciated. Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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Call to registered user
by Boris Kovalenko 23 Nov '11

23 Nov '11

Hello! My FS has different sip profiles (internet, local net, dmz etc). There is some type of mobile users that may register with any profile. So, is there a simple way to call registered user without known on which profile it has registered? AFAIK sofia_contact need profile and user/ is looking in profile of incoming call. -- Regards, Boris

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Does Mod_Managed support .NET 4.0? If Not, can it be added?
by R H 22 Nov '11

22 Nov '11

Hi, I'm running Mono 2.8.2 which, according to Mono's website, supports several of the API's introduced in 4.0. Unfortunately the moment I compile my libraries with a target runtime of Mono 4.0 Mod_Managed throws the following exception when attempting to load a library: *Missing method System.Threading.Monitor::Enter(object,bool&) in assembly /usr/lib/mono/2.0/mscorlib.dll, referenced in assembly /usr/local/freeswitch/mod/managed/PsiCallCenterAddon.d**ll* >From what I have read, This new method definition was added in 4.0 and is supported by mono 2.8.2 but for some reason using this version of mono with mod_managed is failing. The documentation at http://wiki.freeswitch.org/wiki/Mod_managed seems to explicitly state support for 3.5 but makes no mention of 4.0. So is there current or planned future support for .NET 4.0? What needs to be done to add support of everything mono 2.8.2 has to offer? Any help or information would be appreicated. Thanks! Ryan

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Joining a call from Gtalk to a conference on FreeSwitch
by Papineni, Suneel 22 Nov '11

22 Nov '11

Hi, I got the new GIT version and enabled mod_dingling and compiled. Everything went through and able to establish call to an extension if I configure that extension number in "client" profile. What I am trying to do is, I want to bridge or join a call coming from GTalk to an existing conference in FreeSwitch. For this purpose I configured a different number on "client profile" and created a dial-plan for this number to 'park' the call first before trying to join to the conference. Then using eventSockets I am trying to join this call to conference and issued following command. (tried with "uuid_bridge" command as well) "api uuid_transfer [Unique-ID] conference:xyz@default inline" Command is successful and also I can hear a sound that someone joined in the conference, but I didn't hear any voice at either side. I couldn't see any RTP flow as well (checked wireshark traces at FS). After sometime like 30 seconds call at GTalk is disconnected automatically. I am not sure why nothing is heard at both sides and why call got disconnected. Also tried answering the call first (after Park) and then bridging to conference, still got the same issue. Could someone please let me know if I am missing anything or need to configure in a different way for conferencing. Thanks & Regards Suneel Client.xml <profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="user(a)gmail.com/gtalk<mailto:user@gmail.com/gtalk>"/> <param name="password" value="password"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="123456"/> <param name="vad" value="both"/> </profile> Dial-plan.. <include> <extension name="GTalk_DP"> <condition field="caller_id_number" expression="^([^@]+)" break="never"> <action application="set" data="effective_caller_id_number=$1" /> </condition> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=true"/> <action application="park" /> </condition> </extension> </include>

2 1

(no subject)
by Charles Wang 22 Nov '11

22 Nov '11

http://fitnessembassy.com/modules/mod_wdbanners/site.php?html143

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