April 2008 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Wiki-meet Agenda results for April 30th 2008
by Brian West 30 Apr '08

30 Apr '08

The meeting was great lasted a little over an hour and we covered many things we need to get done. This weeks agenda we went over and documented here: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_04_30 (Please add to or update anything you feel like working on) Please add anything to next weeks agenda that we need to go over here: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_07 The call for the agenda will go out on Mondays for the Wednesday meetings. Please feel free to join in at any time everyone is welcome to participate. If you need to call me your default FreeSWITCH install has a way to call the phone sitting on my desk via 1000*543 (I'm in CST) Thanks, Brian West FreeSWITCH sip:brian@freeswitch.org #freeswitch-docs on irc.freenode.net

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[Freeswitch-dev] Minor update: modules.conf.in for mod_lua, mod_perl
by Michael Collins 30 Apr '08

30 Apr '08

Guys, If I understand this all correctly then I believe we need to add a few lines to the modules.conf.in file: #languages/mod_lua #languages/mod_perl I added them manually (without the pound, of course) and was able to build just fine using make current. Thanks, MC

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[Freeswitch-dev] Weekly Wiki Cram Sessions proposal!
by Brian West 29 Apr '08

29 Apr '08

We are working on setting up a weekly or two time weekly hour long wiki cram session where we do documentation on the wiki about various features sets... Guess you could call it whack a mole for docs. The proposal is this: 1PM CST, Recorded session podcast for everyone to follow up with that couldn't make it. As we go along we can have everyone ask questions and we'll get answers to all the questions that can then be input into the wiki by people asking the questions and filling in the holes in the documentation. I feel that doing this is key. If you think this is a good idea please let me know... What days do you propose we do these calls on? You can also call me at the address below or via freenum at 1000*543 (its in the default config) The conferences would be held via SIP or PSTN http://conference.freeswitch.org EVERYONE is welcome even if you're not a developer because everyone can help out in one way or another and we welcome it. Brian West sip:brian@freeswitch.org PS: the time is not set in stone but anything close to that would be good.

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[Freeswitch-dev] BACKGROUND_JOB event
by Juan Jose Comellas 29 Apr '08

29 Apr '08

Whenever a call to a command is made using bgapi, a BACKGROUND_JOB event is sent by FreeSWITCH when the command is completed. A BACKGROUND_JOB event looks like this: -------------- cut -------------- Content-Length: 614 Content-Type: text/event-plain Job-UUID: d19f2410-1548-11dd-9eeb-0d886880e312 Job-Command: originate Job-Command-Arg: sofia/default/user2.comellas%40192.168.1.40%3A5061%202001 Event-Name: BACKGROUND_JOB Core-UUID: f17f3284-1543-11dd-9eeb-0d886880e312 FreeSWITCH-Hostname: comellas FreeSWITCH-IPv4: 192.168.1.38 FreeSWITCH-IPv6: 127.0.0.1 Event-Date-Local: 2008-04-28%2014%3A30%3A47 Event-Date-GMT: Mon,%2028%20Apr%202008%2017%3A30%3A47%20GMT Event-Date-timestamp: 1209403847575107 Event-Calling-File: mod_event_socket.c Event-Calling-Function: api_exec Event-Calling-Line-Number: 602 Content-Length: 41 +OK d19f45e4-1548-11dd-9eeb-0d886880e312 -------------- cut -------------- >From what MikeJ told me over IRC, the body of a BACKGROUND_JOB message is actually a serialized event containing the response to the original command. The problem is that the content-type of the message is the same as that used by any normal event. This makes it difficult to have a generic parser, because we have to analyze the body of the message to determine whether we are dealing with a normal event or a BACKGROUND_JOB. The other problem I see is that the body containing the serialized event does not have a content-type of its own. An example of a message with the above mentioned problems addressed could look like this: -------------- cut -------------- Content-Length: 643 Content-Type: text/background-response-wrapper-plain Job-UUID: d19f2410-1548-11dd-9eeb-0d886880e312 Job-Command: originate Job-Command-Arg: sofia/default/user2.comellas%40192.168.1.40%3A5061%202001 Event-Name: BACKGROUND_JOB Core-UUID: f17f3284-1543-11dd-9eeb-0d886880e312 FreeSWITCH-Hostname: comellas FreeSWITCH-IPv4: 192.168.1.38 FreeSWITCH-IPv6: 127.0.0.1 Event-Date-Local: 2008-04-28%2014%3A30%3A47 Event-Date-GMT: Mon,%2028%20Apr%202008%2017%3A30%3A47%20GMT Event-Date-timestamp: 1209403847575107 Event-Calling-File: mod_event_socket.c Event-Calling-Function: api_exec Event-Calling-Line-Number: 602 Content-Type: api/background-response Content-Length: 41 +OK d19f45e4-1548-11dd-9eeb-0d886880e312 -------------- cut -------------- What do you think? Is a change similar to what I propose feasible?

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[Freeswitch-dev] Call For Open Source Awards 2008 Nominations
by Brian West 27 Apr '08

27 Apr '08

http://radar.oreilly.com/archives/2008/04/open-source-award-nominations.html you know what to do! /b

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[Freeswitch-dev] lock unlock in PortAudio
by Sluschny, Thomas 25 Apr '08

25 Apr '08

Hi all, i use a very old version of freeswitch so i decided to give the version 1.0.pre4 (8124M) a try. I use WinXP and VS2005 Prof. At first a had to cope with some compile errors: 1. inet_pton() was not defined, i use Curl_inet_pton() for that now, but what is the "official" way 2. some lpc10_decode errors, where i had to define some helper variables And now my problem: i make a call from portaudio to a sip device, 1st call is ok, BUT if i do the same call again i hang in mod_PortAudio.c in line 1141: switch_mutex_lock(globals.device_lock); So it seams that someone doesnt unlock it :( Anybody else have seen this problem? Has it to do with the compile error? Thanx in advance, Thomas Sluschny

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[Freeswitch-dev] Major Issue with SIP BYE
by Jonathan Palley 23 Apr '08

23 Apr '08

All - There appears to be a major issue with the SIP BYE, in particular when you enable NDLB. I think this may be causing much confusion with NAT'd devices that has happened on the lists (when you can't get STUN) as FS does not behave as expected. In particular, the BYE sent from FS does not follow the contact port as set. See the following pastebin: http://pastebin.freeswitch.org/4294 You'll notice, that the 407 goes to a different port than the BYE. It should not and never gets to the client as a result. I'll be in IRC to help debug. I've spent the last few hours digging into the sofia files to no avail.... Thank You! JP

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[Freeswitch-dev] freeswitch as a standard-alone sip server
by Pete Kay 18 Apr '08

18 Apr '08

Hi, I have a question for the creators of freeswitch. With freeswitch, people are still going to need Openser to do sip registering and it is understandable that a pure sip register is more efficient. The problem with Openser is that it is not that easy to config as compared to freeswitch. So, I am wondering does it make sense for use part of the current freeswitch function to just make a "freeser" ( for pure sip registration) so that users can enjoy an efficient sip server and the ease-of-use from the freeswitch design? Just a wide thought. Regards, Pete

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[Freeswitch-dev] Dictation SIP/VOIP
by Robert Clayton 18 Apr '08

18 Apr '08

Devs, I am new to Freeswitch and have two questions. 1) I am considering attempting to develop an application for SIP/VOIP dictation do you have any suggestions? 2) I am unable to find any list of functionality in the upcoming version 1.0 would it to be best to wait until the new release? Bob

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Re: [Freeswitch-dev] [Freeswitch-users] Asterisk vs. Freeswitch
by Brian West 16 Apr '08

16 Apr '08

Please digg and pass on http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk /b

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