March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Event-List: double Content-Lenght?
by Iñaki Baz Castillo 22 Mar '11

22 Mar '11

Hi, I'm trying to understand the following example in the doc: http://wiki.freeswitch.org/wiki/Event_List Upon job completion, server response ---------------------- Content-Length: 625 Content-Type: text/event-plain Job-UUID: 7f4db78a-17d7-11dd-b7a0-db4edd065621 Job-Command: originate Job-Command-Arg: sofia/default/1005%20'%26park' Event-Name: BACKGROUND_JOB Core-UUID: 42bdf272-16e6-11dd-b7a0-db4edd065621 FreeSWITCH-Hostname: ser FreeSWITCH-IPv4: 192.168.1.104 FreeSWITCH-IPv6: 127.0.0.1 Event-Date-Local: 2008-05-02%2007%3A37%3A03 Event-Date-GMT: Thu,%2001%20May%202008%2023%3A37%3A03%20GMT Event-Date-timestamp: 1209685023894968 Event-Calling-File: mod_event_socket.c Event-Calling-Function: api_exec Event-Calling-Line-Number: 609 Content-Length: 41 +OK 7f4de4bc-17d7-11dd-b7a0-db4edd065621 ---------------------- What is the purpose of the first "Content-Length: 625"? Why not just send all the headers with no Content-Lenght? (just leave the second Content-Length:41 which is the length of the job uuid). So what is the purpose of the first Content-Length? A parser doesn't need it at all as it can read headers until it finds "Content-Length" (so it reads N bytes more) or it finds a double \n\n (so event info is terminated). Thanks for any clarification. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Gawk.exe crashes when building on Windows 7 64bits w/ VS 2008
by Frankie Yiu 22 Mar '11

22 Mar '11

Hi there, I am wondering if anyone else encounters the same issue. I am running Windows 7 64bits and when I build FreeSwitch in Visual Studio 2008 (SP1) for WIN32, it crashes/compilation error when opening a Gawk program which causes files can not be automatically generated. I even installed a newer version 3.1.7 and it still crashes. Please let me know if you have resolved this issue. Thanks, Frankie

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How can I send an event via SIP from one FreeSWITCH to an other FreeSWITCH?
by Чистяков Иван 22 Mar '11

22 Mar '11

How can I send an event via SIP from one FreeSWITCH to an other FreeSWITCH?

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send custom tcp message from esl
by Ejay Greeves 22 Mar '11

22 Mar '11

I am using esl in Ruby I want to send tcp messages to a server. When I tried to make my tcpsocket class it did not work it must be messing with the socket setup of esl. Is there a way without the need to manage the sockets seperatley to use els methods to send out custom tcp message. All I need is for a simple string to be fired to tcp port. How can I do this from inside esl.

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ClueCon 2011 Registration Open, Call For Speakers
by Michael Collins 22 Mar '11

22 Mar '11

Greetings fellow open source VoIP users and developers! We are happy to announce that ClueCon 2011 registration is now open. The event will be held August 9-11, 2011 at the Sofitel Hotel in downtown Chicago. Additionally, we are now accepting proposals for speaking topics. If you or your organization would like to speak at ClueCon this year please contact us with your ideas. Visit us at http://www.cluecon.com for additional details on registration, speaking, or sponsorship opportunities. Thank you to all the members of the open source telephony projects who have supported ClueCon over the years. We hope to see you all again this coming August! Michael S Collins ClueCon Team Member http://www.cluecon.com

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New FreeSWITCH Module: mod_ladspa
by Michael Collins 22 Mar '11

22 Mar '11

Hello all! The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story: http://www.freeswitch.org/node/313 Have fun! -Michael

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Lua playAndCollectDigits timeout problem. Possible bug
by Pete Kelly 22 Mar '11

22 Mar '11

Hi, I am trying to invoke a timeout using the session:playAndCollectDigits lua function, this is the syntax I am using: call_number = session:playAndGetDigits(1, 20, 1, 10000, "#*", "some-file.wav", "", "[0-9]{1,}"); If the user doesn't hit * or # I need the function to return after 10 seconds. However this timeout seems to only kick in if I start entering DTMF *after* the initial prompt has played. If I start entering DTMF during the prompt, the function never returns due to timeout. Has anyone else seen/experienced this ? Pete

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Does Freeswitch complies with SIP Connect technical recommendation?
by Yehavi Bourvine 22 Mar '11

22 Mar '11

Hello, We published an RFP for our SIP connection to the PSTN. One of the suppliers asked whether Freeswitch complies with the SIP CONNECT recomendation. I have to answer them soon, and don't have enough time to read and evaluate the entire document at present. Does Freeswitch fullfills all the recomendations there? Thanks, __Yehavi:

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More then one number for registered user
by Boris Kovalenko 22 Mar '11

22 Mar '11

Hello! Is this possible to have one registered user and more then one associated number with it (without tricks with dialplan)? For example: <user id="XXX" number-alias="YYY,ZZZ....."> -- Regards, Boris

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I need to register all of my DIDs with a single SIP trunk..
by Randy Andrade 22 Mar '11

22 Mar '11

Greetings, I'm hoping someone can help me out here. I'm relatively new to FS, but not to general telephony and I have a small amount of SIP experience. My FS installation is, so far, a fairly straightforward PBX setup with a single SIP trunk with 21 DIDs to the ITSP that I work at. I have everything working as far as inbound and outbound calls, except that in order for me to use a DID as the outbound caller id, that DID needs to be explicitly registered with the ITSP, and not just the pilot number for the SIP trunk. When I try to create a second gateway *.xml file to register a second DID, I get an error "Registration Failed with status Operation has no matching challenge [904]" on the FS console. As I mentioned, I work at this ITSP and I program similar SIP trunks into Adtran TA9xx boxes for SIP over T1 deliveries. On the Adtrans, I'm able to define a single SIP trunk, and have multiple register statements within that trunk, one for each DID. I'm wondering if there's a way to do something similar in FS. I'm running a very recent (I update from the GIT tree about twice a week) version of FS on a Debian Linux OS, and, I have several local extensions working and able to call in and out; I just want to be able to outpulse various DIDs as my caller-id without getting calls rejected. Here's a copy of my gateway .xml file, with relevant information changed: <include> <gateway name="gatewayname">  <param name="username" value="1234567890"/>      <param name="from-domain" value="gatewayname"/>  <param name="password" value="XXXXXXXX"/>    <param name="proxy" value="proxyname"/>      <param name="register" value="true"/>      <param name="caller-id-in-from" value="true"/>    <param name="ping" value="30"/> <param name="extension-in-contact" value="true"/> </gateway> </include> Again, I don't know if I would do it here (I would assume so) but I'm wondering if there's some way (maybe multiple "extension" statements? I dunno..) to register multiple DIDs in the same gateway / SIP trunk. Thanks!

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