July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Command to get FreeSWITCH's internal time
by Milena 27 Jul '09

27 Jul '09

Hello everyone, I'm using the inbound event socket to receive some information about the status of my FreeSWITCH system and i wanted to know if there is an api command that can be used to get the FreeSWITCH time, I tried searching around in the docs and in google but i couldn't find an answer. Thanks for your attention and thanks in advance if anyone can assist me with this. Have a nice time and lots of cookies :)

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Slashdot: How-to-Help-With-a-University-ICT-Strategy
by Giovanni Maruzzelli 27 Jul '09

27 Jul '09

http://ask.slashdot.org/story/09/07/27/1652247/How-to-Help-With-a-Universit… " An anonymous reader writes "I have been asked to contribute to my university's revised ICT (Information and Communication Technology) strategy and I am curious what fellow Slashdot members consider to be the main advice in this context. What are the major mistakes that organizations like universities make? Given the complexity of the different participants in a university, how does one have a coherent strategy that fulfills the needs of such a wide audience? How does one promote open source in a managerial culture? How does one deal with the curse of the virtual learning environment?"" http://ask.slashdot.org/comments.pl?sid=1316571&cid=28842157

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Which method Can I use in IVR
by Thangappan.M 27 Jul '09

27 Jul '09

Dear all, I am learning how to implement a IVR in Freeswitch.In our organization we are using Perl scripting language for doing this.So In freeswitch also I need to use Perl. So far I heard two methods for executing IVR. One is in dial plan using perl application.( In perl I create IVR menu and play the voice files) Another one is using event socket.In dial plan I specified socket application and write a Perl script which is listening that particular port and get the session Id. Have I understood correctly?.If it is correct means tell which method can I use?. Other make me understand well. I have seen downloaded perl IVR menu from freeswitch site.In that they called some internal functions like playandGetDigits,StreamFile,ready ...etc. These functions is been called by using $session variable.Where these functions are defined.? -- Regards, Thangappan.M

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ClueCon 2009 - Last Chance For Early Bird Special!
by Michael Collins 27 Jul '09

27 Jul '09

ClueCon is next week! We're all gearing up for a great event. Here is some important information for those who've not already paid: Today is the last day to receive the $499 early bird special. After today, the price will go up to $699. If you have registered at the ClueCon website but you have not yet paid then please call 877.742.CLUE immediately! We want to make sure that you get the early bird rate. All registrations after today (Monday July 27) will be $699. Thank you for your support of ClueCon 2009! We are looking forward to seeing everyone in person in Chicago. -Michael Collins http://www.cluecon.com 877.742.CLUE

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freeswitch and siremis v0.9.3
by Daniel-Constantin Mierla 27 Jul '09

27 Jul '09

Hello, recently released siremis v0.9.3 adds support for communication with freeswitch via event socket. siremis is an open source web management interface targeting the SIP routing engines kamailio (openser) and sip-router.org. freeswitch fits perfectly in the picture since it completes the routing engines with rich media services. The new release includes php code to communicate with freeswitch via tcp/event socket and a panel to send commands/display response. Code is grouped like a library, new features being straightforward to develop. For some commands, the output is pretty formatted - screenshot: http://www.asipto.com/gallery/v/siremis/siremis_20.jpg.html?g2_imageViewsIn… More is planned for the future (e.g., display active calls of a certain user, click to end an active call). Feedback and contributions are welcome, visit: http://siremis.asipto.com Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/

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Limit is not working when originate a call
by Eli Hayun 27 Jul '09

27 Jul '09

Hi I set the limit to 1 on the extension like that <action application="limit_hash" data="${destination_number} ${destination_number} 1" /> When I am trying to make a call the that destination i transfered to limit_exceeded dialplan, just like I want The problem is, that when I am trying to make a call using "originate" I am not getting the limitation. Why is that?

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core dump
by Baskar 27 Jul '09

27 Jul '09

Hi, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9851 -- Thanks with Regards, N.Baskar

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Question: Capturing VoiceData received from Sagnoma E1 card
by Helmut Kuper 27 Jul '09

27 Jul '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, For outgoing calls I'm hunting the cause for missing some 100ms of voice data send from remote right after pickup the remote phone (e.g. initial "Hello?" sound like "o?" or even nothing) On FreeSwitch server I captured the VoIP data to the called VoIP-Phone on the sofia interface. Using wireshark it also shows that the voice data from remote is missed. Using Mobil phones or ISDN phones calling the same remote party there is never a bit missed. This problem occurs rare - once or twice per day and per local voip phone, but it's quite anoying. So is there a way to capture the correspondig ISDN voice data FS receives before it is transmitted via RTP or just droped? I want to c whether FS drops the early RTP packets or whether FS never got the data from ISDN. Sofia Profile is using <param name="inbound-late-negotiation" value="true"/> The dialplan portion is: <extension name="outgoing-pstn"> <condition field="destination_number" expression="^94([0-9]+)$" break="never"> <action application="privacy" data="full"/> <action application="set" data="effective_caller_id_name=anonymous"/> <action application="set" data="effective_caller_id_number=anonymous"/> </condition> <condition field="destination_number" expression="^([0-9]+)$"> <action application="set" data="ignore_early_media=true"/> <action application="set" data="absolute_codec_string=PCMA"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="openzap/1/a/$1"/> <action application="transfer" data="${destination_number} XML et_internal_error"/> </condition> Any ideas to refine my debugging? regard helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKZcpS4tZeNddg3dwRAnVDAKCxXXkdbf0RKeeSMFYucCIno3tA9gCfUzbD 148BfuKavTtBoJNScRQDmSk= =JbtY -----END PGP SIGNATURE-----

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Multiple gateways: Priority the first bridge with prefix
by Edmar Cruz 27 Jul '09

27 Jul '09

Hi FS Users, I just want to try multiple gateways. It works actually like this... <action application= "bridge" data="sofia/sip/51(a)222.333.444.555 | sofia/sip/63(a)111.222.333.3333"/> But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Multiple gateways: Priority the first bridge with prefix
by Edmar Cruz 27 Jul '09

27 Jul '09

Hi FS Users, I just want to try multiple gateways. It works actually like this... <action application= "bridge" data="sofia/sip/51(a)222.333.444.555 | sofia/sip/63(a)111.222.333.3333"/> But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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