November 2011 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Other-Leg-uuid before bridge
by succer110＠tiscali.it 24 Nov '11

24 Nov '11

Hi everybody! I'm executing a script with exec_on_answer and exec_on_pre_answer in a bridge application. Since the execution of my application is triggered before bridging the two legs (that's exactly hat what i was looking for) i do not have the "Other-Leg-uuid" var setted in my session's channel variable. Is there any way to get the uuid of the bleg before the bridge? <action application="export" data="nolocal:execute_on_pre_answer=lua dial.lua PRGS"/> <action application="export" data="nolocal:execute_on_answer=lua dial.lua ANSW"/> <action application="bridge" data="sofia/external/111xxxx(a)192.168.1.1"/> So: in the dial.lua i need to have the uuid of "sofia/external/1111(a)192.168.1.1". Thank you in advice! (and thank you all for freeswitch!!) E' nata indoona : chiama, videochiama e messaggia Gratis. Scarica indoona per iPhone, Android e PC

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 23 Nov '11

23 Nov '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_23 There have been a number of changes/additions to mod_conference, so we are going to talk about those today. Also, there are several FreeSWITCH updates to talk about. If we have time and if enough subject matter experts are available we will also talk about SIP presence. See you soon! -Michael

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[Freeswitch-dev] Submitted a patch to correct compile error of mod_shout inFC 15 & 16
by Joegen Baclor 22 Nov '11

22 Nov '11

Jira is here http://jira.freeswitch.org/browse/FS-3711

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[Freeswitch-dev] why does app "record_session" record extra frequency range
by Han Xin 22 Nov '11

22 Nov '11

Hi all, I have a bridged call and a home brew module which adds a media bug to the local and partner session before the bridge app in the dial plan. The original speech codec used is G722, which sampling rate is 16KHz. I add a media bug which does low pass filtering the speech to 4KHz. But the spectrum of the recorded .wav file showed that it always contains the freq range from 0K to 8K. The desired spectrum should contain only 0 to 4KHz. I am confused, can anyone show me some hints? Thanks in advance. Best Regards, Han --

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[Freeswitch-dev] Wednesday Conf Call - SIP Presence?
by Michael Collins 22 Nov '11

22 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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[Freeswitch-dev] Connect Voice mail flow before sending Answer.
by Sanath Prasanna 18 Nov '11

18 Nov '11

Hi all, I am new to Freeswitch & need to do following. When called party not available, I need to route call to voice mail system without sending answer.(or OK) If send answer, calling party will be charged. So I need to avoid that. After that I need to prompt message as "Press 1 for go to voicemail or Press 2 to exit" If user press1 then send answer (Charging is start) & connect to voice mail call flow. For doing above, shall I need to do code level changes ?? Pls advice. Br, Sanath

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[Freeswitch-dev] Loading an IVR menu on the fly through an URL
by RaviRaj Mulasa 18 Nov '11

18 Nov '11

Hi FreeSWITCH Enthusiastics/Gurus Requirement 1. Customers can build a custom IVR using a UI. 2. Back end code persists the IVR details in a DB. 3. A URL will be exposed and will return the a XML file in the format as follows <menu name="CUSTOMER_ID_ivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="/mnt/callControlStorage/en/invalidSelection.wav" exit-sound="/mnt/callControlStorage/en/tollFreeForwardingDotCom.wav" tollFreeForwardingDotCom timeout ="10000" inter-digit-timeout="2000" max-failures="3" digit-len="4"> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888(a)conference.freeswitch.org"/<mailto:sofia/$$%7bdomain%7d/888@conference.freeswitch.org%22/>> <entry action="menu-exec-app" digits="2" param="transfer 9996 XML default"/>  <entry action="menu-exec-app" digits="3" param="transfer 9999 XML default"/>  <entry action="menu-sub" digits="4" param="3107767546_ivr_submenu"/>  <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>  <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML default"/>  <entry action="menu-top" digits="9"/>  </menu> 4. Load and start executing the IVR 'CUSTOMER_ID_ivr' from events socket on the fly via HTTP URI. Gave it a shot with current setup ivr.conf.xml <configuration name="ivr.conf" description="IVR menus"> <menus> <X-PRE-PROCESS cmd="include" data="/mnt/<<Folder Name for the NETWORK DIRVE mapped>>/ivr_menus/*.xml"/> </menus> </configuration> The ivr.conf.xml is auto loaded at startup of FreeSWITCH. To load any new IVR menus , we need to call API command reloadxml which reloads the whole XML configs in autload_configs folder. The reloadxml might take long time based on the number of XML files present in the folder "/mnt/<<Folder Name for the NETWORK DIRVE mapped>>/ivr_menus" Good /Hope to have for the IVR menu 1. Load /Unload IVR menus on the fly using API command(s) freeswitch@localhost> loadivr << URL to fetch the XML describing the IVR menu(s)>> freeswitch@localhost> unloadivr <<IVR menu name>> 2. From the event socket SendMsg <uuid> call-command: execute execute-app-name: ivr execute-app-arg: <<MAIN_MENU to start execution>>SPACE <<URL to fetch the XML describing the IVR menu(s)>> Educated/Wild Guess We might be able to play IVR menu on the fly by combining the mod_xml_curl and dialplan tool - ivr. Please let us know how can we achieve the functionality , we are more interested in the second approach(Event Socket) under Good /Hope to have for the IVR menu. Thanks RaviRaj.

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[Freeswitch-dev] Joining a call from Gtalk to a conference on FreeSwitch
by Papineni, Suneel 17 Nov '11

17 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"/> <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>

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[Freeswitch-dev] event dispatch thread should exit when idle
by john 17 Nov '11

17 Nov '11

Dear Friends, I was going through the code of event mechanism in freeswitch. In switch_event.cThe switch event thread may spawn more dispatch threads in case heavy load.But I am wandering why there isn't any exit mechanism of the event dispatch thread. event dispatch thread only exits when shutdown. Current event dispatch queue size is 10,000. At some point of time event thread detects 12,000 events to be dispatched. It spawns a new dispatch thread - 2, to deliver remaining 2000 events. but after that dispatch thread -2 remains active & idle. I think we need to kill this idle thread+queue, so that it can be re spawned latter on. and max no of dispatch threads never get exceed. Please reply me if anyone has implemented this or have any idea abt this. thnx

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[Freeswitch-dev] How can I detect the cause of a potential memory leak?
by François Delawarde 17 Nov '11

17 Nov '11

Hello, I'm experiencing what I think is a memory leak in a production system with a recent git (2-Nov), with FS consuming >5GB and increasing after a few days with never more than 10 calls at once. While the users are not experiencing problems so far, I'm a bit worried. So I have a few questions: - How can I be sure it's a leak and not some memory pool thing that FS would free when the system needs (system has 10GB total)? - If I unload modules one by one, is the memory used by this module freed immediately? - What would be a "good" way to try and narrow down the cause? Is valgrind a good tool for that? Of course I'll try with GIT HEAD and try my best to find the cause before considering to add a new Jira issue, otherwise it's quite useless. Thanks, François.

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