December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

(no subject)
by veerabhadrarao` 26 Dec '12

26 Dec '12

hai I am using Freeswitch 1.2.4 currently i am working on Attendent transfer can u please help me how to make attendant transfer. let when i make a call A--->B and B-->C then how can i connect to A--->C please help me

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FS opening ports
by stephen＠picardogroup.com 26 Dec '12

26 Dec '12

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FreeSWITCH Community Conference Call Today
by Michael Collins 26 Dec '12

26 Dec '12

Hi gang! I know a lot of folks are gone for the holidays but anyone who is around is welcome to join us today. We have a light agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_26 However, I believe at least one person will be joining us to talk about a potential job for a FreeSWITCH contractor. Hope to talk to you all soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Creating a call app based on freeswitch (and adhearsion)
by Nathan Samson 24 Dec '12

24 Dec '12

Hi all, We have a lot of customers that should be able to call our information center. Based on who (and when) he calls, the call should be forwarded to one specific person (or stored as voice message outside the hours). Since forwading the call to a mobile (or fixed) phone is costing money, we want to forward it to SIP clients when this contact person is available via SIP (so we first try SIP, if the contact person doesn't take the call we'll try the phone number). We also want to do outbound calls (as internal users), some via a normal phone (if internet is not available), or also via SIP. We also have a admin application ourselves to manage users, information stored in this database should be used to specify the incoming calls forwarding. The main question is how we best manage our SIP users (we prefer to do everything in our application). Are there any APIs available to create/delete & manage SIP users for freeswitch? Can we write (in a reasonably short amount of time) an authentication plugin for freeswitch which will consult our internal database. Note that our application logic (how things should be forwarded, what application should run etc) will be written in the adhearsion framework, so every call should be forwarded to adhearsion, which wil lthen handle all logic (except for the SIP authentication). Cheers, Nathan

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Re: [Freeswitch-users] xml_curl and ivr menus
by Dave May 24 Dec '12

24 Dec '12

Sent from my HTC smartphone on the Now Network from Sprint! ----- Reply message ----- From: "Tim Meade" <Tim.Meade(a)Millicorp.com> To: "FreeSWITCH Users Help (freeswitch-users(a)lists.freeswitch.org)" <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] xml_curl and ivr menus Date: Mon, Dec 24, 2012 2:27 pm I am continually getting 'unable to find menu' while trying to retrieve an ivr_menu using xml_curl Using xml_curl debug_on My DialPlan <document type="freeswitch/xml"> <section name="dialplan" description="example_curl_dialplan"> <context name="public"> <extension name="ws_origination"> <condition field="destination_number" expression="^\d+$" break="on-true"> <action inline="true" application="set" data="continue_on_fail=true"/> <action inline="true" application="set" data="hangup_after_bridge=true"/> <action application="answer"/> <action application="ivr" data="tivr"/> </condition> </extension> </context> </section> And the ivr configuration returned by my xml_curl <include>   <menu name="tivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="rms" confirm-attempts="3" timeout="10000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="4">   <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888(a)conference.freeswitch.org"/> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/>  <entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/>  <entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/>  <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 features"/> <entry action="menu-top" digits="9"/>  </menu> </include> Any ideas on the 'unable to find menu' ???? Thanks Tim

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Performance hit originating calls via event socket
by Muhammad Naseer Bhatti 24 Dec '12

24 Dec '12

WIth the latest development head running on a Debian box, I am trying to originate calls using event socket. Not sure why, but not able to generate more than 700 calls at any given time. Roughly around 40 CPS. For the sake of testing and simplicity, I just put the whole originate command multiple times in a text file and looping around it with fs_cli to execute. Dialplan sends the call to a lua script which randomly answers the call run the echo application and terminate with random seconds, something like http://wiki.freeswitch.org/wiki/Fakecall_responder CPU remains low all the time. Looks like the opening and closing of ports to esl is the reason slowing originate rate? I have tried from 8 to 24 CPU cores but the behavior remains the same. FS-4962 opened already in case we want to proceed from there. Thanks, -- Muhammad Naseer Bhatti

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MWI-Account - Issues
by Lloyd Aloysius 24 Dec '12

24 Dec '12

Hi All: I am trying to send a user MWI to a different registered user *Eg: user : amirt .. directory parameter is like below* <param name="MWI-Account" value="amrit(a)mydomain.com"/> Every thing works as expected. ============================================================ Now I change the MWI-Account to a different user *user : amirt .. directory parameter is like below* * * <param name="MWI-Account" value="eddie(a)mydomain.com"/> Now user: amirt phone stop MWI . That is correct. But the user eddie device not Light up for the user : amirt voice mail notification I have the following in the sip profile <param name="send-message-query-on-register" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> Am I missing something. Thanks Lloyd

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mod_sofia gateway proxy definition, rtmp voice codec
by bratner bratner 22 Dec '12

22 Dec '12

Dear List, I'm seeking an answer to the following questions: 1. If I define a gateway under a sip profile and I do it with "outbound-proxy" setting then the INVITE message to that proxy will also include the proxy in its RURI. The expected behavior is that the message will be sent to to the specified proxy BUT the RURI will point to the "domain" as defined in the gateway config. That behavior is also correct for REGISTER request and this is the way it actually works if I define "register-proxy". REGISTER request go to the ip:port of the defined "register-proxy" but the domain part in RURI will be set to the domain from gateway configuration or "register-domain". 2. mod_rtmp is hardcoded with SPEEX. So is of course the flex client. Is there a reason for this? According to flex docs, PCMU/A are also supported. I also heard that the module is not really maintained. Regards, bratner.

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Problem in DTMF detection with PRI ( Duplicate DTMF / DTMF Removal )
by ganesh kumar 22 Dec '12

22 Dec '12

Hi All, We are using the following version of FreeSWITCH. Version : FreeSWITCH Version 1.0.head (git-8f2ee97 2010-12-05 17-19-28 -0600) Linux Kernel Version : Linux My-Host 2.6.35-22-generic-pae #35-Ubuntu SMP Sat Oct 16 22:16:51 UTC 2010 i686 GNU/Linux Sangoma PRI Card : AFT A102 Wanpipe Version: WANPIPE Release: 3.5.19 Libsng Version: libsng_isdn-5.3.0.i686 I am having an application to receive DTMF and process based on the received DTMF. The problem is, whenever receiving DTMF some of the digits got duplicated and sometimes some digits are missing. I have searched lot about this problem. And I came to know that many people faced this issue. But I couldn't get proper solution. I want to know whether this problem is solved in freeswitch or not. If it is solved, what is the solution or workaround for this problem. Note: I have tried disabling the core DTMF detect and enabled the spandsp DTMF detect but it too failed some times and it is not a reliable solution for my problem. Kindly help me out. Thanks in advance. -- Regards, Ganeshkumar.S

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Allow calls from Gateway based on GW IP Address
by Sean Devoy 22 Dec '12

22 Dec '12

HI All, I am trying to setup Vitelity as a FS Gateway. The wiki is 2 years out of date, the menus @Vitelity have all changed and I have been unable to get the FS setup to work. Vitelity recommends IP Based Authentication. I have read the wiki regarding this and I am simply thrashing around. So, here is the pastbin of the global siptrace: http://pastebin.freeswitch.org/20354 Sorry about all the extra line feeds, they are not visible in notepad! I just want to know how to setup FS to all accept all calls from a FQDN or IP Address and route them to a specific context (like "from-trunk"). I will pay the WIKI TOLL when we get this working. Thanks, Sean

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