March 2008 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

Announcing the OpenZAP.hps project
by Hans Petter Selasky 24 Mar '08

24 Mar '08

Hi, During the recent week I have been talking to the people at FreeSwitch.org and we have agreed to build up a new full featured telephony stack by merging ISDN4BSD and OpenZAP. I have called the new branch OpenZAP.hps in my SVN repo. As a consequence of this I have therefore decided that the ISDN part of my ISDN4BSD project will only receive bug-fixes from today and on, and will be considered a stable branch. Some words about what OpenZAP.hps will bring. The fundamental part of the new telephony stack is the node concept. A node is a part that can receive "mbufs" using FreeBSD terms. There will be created different node types like echo-cancel, format conversion, tone detection, DSS1 protocol, CAPI2.0, ... Nodes are addressed by a 32-bit value. Everything will be built around the node concept. This gives us an important advantage that nodes can be placed in userland, kernel or even at a remote host. In other words the load can be distributed easily. Multithreading. Unlike many other telephony implementations, there will not be one thread for every B-channel or call present. Instead there will be one thread for each CPU and possibly one configuration thread running at a lower priority. Compatibility. The new stuff will be backwards compatible with CAPI2.0 and possibly also the FreeSwitch API, which means you will still be able to use it with Asterisk and Callweaver through "chan_capi", and FreeSwitch through "libopenzap". Planning. I suspect that OpenZAP.hps will be ready for testing before Christmas of 2008. Financial support. Feel free to support the project economically. If you are interested in working for the project, and have the right experience, there is a chance that something can be arranged so that you will get some money in return. Currently I cannot offer any full-time positions. --HPS References: ISDN4BSD repo: ============== svn --username anonsvn --password anonsvn checkout svn://svn.turbocat.net/i4b/trunk/i4b OpenZAP.hps repo: ================= svn --username anonsvn --password anonsvn checkout svn://svn.turbocat.net/i4b/trunk/openzap.hps

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libteletone multi tone detection usage
by Moises Silva 23 Mar '08

23 Mar '08

Hello, I have been working in a MFC/R2 signaling library that eventually could be used in FreeSwitch, the library is working pretty well for the R2 México variant and it should be easy to add new protocol variants. I am just missing the handling of some tones (congestion, busy, call free of charge) but that will be coming soon. During my testing I have been using spandsp to detect and generate the required 15 MF signals. As you may know, spandsp is GPL and not something I plan to depend on. I want to move from spandsp to libteletone. As a first step I created a small test case thatn generates forward MF tones with spandsp and then try to detect them using both spandsp and libteletone to compare results. The test case is very simple (100 lines), its source code can be found here: http://pastebin.com/m7156bc21 If you run the test case you will see that spandsp can detect all of the 15 MF tones using just 320 samples for each one, however, libteletone, apparently (probably because I am not using it properly) can only detect 9 tones using different large amount of samples (4480, 3520 etc), the remaining 6 tones are not detected at all even after reaching the 8000 samples limit I specified. Can somebody advice about what could be possibly wrong in how I use libteletone to detect MF signals in the test case? Any hints will be appreciated. Kind Regards, Moisés Silva -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire

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Re: [Freeswitch-dev] [Freeswitch-users] Sun T2000 Fund Raiser...
by Matt Porter 23 Mar '08

23 Mar '08

Rent it.. http://www.btinet.net/01_Server_Host/03a_Solaris_SPARC/index.html http://dedicatedserver.com/solutions/hosting/sun.cfm -------------------------------------------------- From: "Brian West" <brian.west(a)mac.com> Sent: Wednesday, March 19, 2008 10:30 PM To: <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Sun T2000 Fund Raiser... > Well the 8 core is about 20k.. but you do 60days you get 25-45% off. > > /b > > On Mar 19, 2008, at 10:24 PM, Vijay Ramnarine wrote: > >> whats the cost like for one of these babies? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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RC1 Has Arrived!
by Anthony Minessale 20 Mar '08

20 Mar '08

FreeSWITCH Community, After nearly 2 years of non-stop development, FreeSWITCH has evolved from an idea and an empty directory into a fully operational soft-switch capable of withstanding the loads 8-10 times as large as anything else they were able to deploy in the past with other software according to the reports from our early adopters. I am happy to announce the immediate availability of FreeSWITCH 1.0 “Phoenix“ (Release Candidate 1). The Phoenix symbolizes the rebirth of a new era and I hope this new release will be the first step in a new age of voice-over-IP development we can all benefit from. We anticipate the official FreeSWITCH 1.0 “Phoenix” release date to be Monday, May 26th 2008. We will have a formal release presentation at this year’s ClueCon 2008 August 5th-7th 2008. http://www.cluecon.com I would like to thank our community for the testing, bug reporting, code contributions, feedback and financial donations that helped us get this far, especially those who were involved from day 1. Please take this opportunity to test drive FreeSWITCH as we will be making it available as binary packages for the first time to practice for the official 1.0 release. The code and arrangement of the configuration will not change apart from bug fixes and tweaks from this point forward. As always, the code will still be available in svn at the usual place. For download instructions please stay tuned to: http://wiki.freeswitch.org/wiki/Installation_Guide The packages will be online soon. Thanks Everyone! Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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CONGRATS ON RC1!!!
by Michael Collins 20 Mar '08

20 Mar '08

-----Original Message----- From: Michael Collins Sent: Wed 3/19/2008 11:26 PM To: freeswitch.org Subject: CONGRATS ON RC1!!! Guys, Nice work on getting RC1 tagged today. Question: I'd like to get the wiki updated with how to snag and install with RC1 - is it okay to tell people to go to this page and grab the appropriate file? http://files.freeswitch.org <http://files.freeswitch.org/> Also, with RC1 is there a particular procedure to follow with the sound files or can the user still do make sounds-install and/or make moh-install? I'll get the wiki updated with all this fun stuff. Lastly, I updated Wikipedia with the correct date for RC1. I accidentally changed it a few months back when I misunderstood something Tony said on channel - we were at beta3-ish and I saw a mention of RC1 and inferred too much. Anyway, WP is now up-to-date. http://en.wikipedia.org/FreeSWITCH. -MC P.S. - It would be helpful to have some sort of official release notice somewhere on the freeswitch.org site, especially something that can be linked-to so that we can add links to WP and to voip-info.org's news page.

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Sun T2000 Fund Raiser...
by Brian West 20 Mar '08

20 Mar '08

Ok guys we are trying to raise some funds to buy an 8 core T2000 box for the FreeSWITCH project. Anyone wishing to assist in this endeavor please email me off list. Thanks, Brian West brian(a)freeswitch.org

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dial plan - event socket question
by jonathan augenstine 19 Mar '08

19 Mar '08

I would like to pass data from the dial plan/mod-java to a Java application on the Event socket interface. Is there anyway to do that through Freeswitch or do I need to pass it via normal Java mechanisms. Jonathan

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Re: [Freeswitch-dev] freeswitch Scenario
by Yogesh Patel 17 Mar '08

17 Mar '08

#1 - Is your test server up and running right now? --> tets server is up and running but its on Private IP. #2 - Are you running with the default configs? --> we are running with default configured, configured wikipbx earlier with freeswitch but removed later when we found it can not help us with what we want. #3 - Have you made any changes or additions to the default configs? --> we have made few changes to make things working and did stuff here and there but nothing removed, and almost everything is same except added some more files. #4 - What revision are you on? --> we use SVN so I guess it will be the latest one. Let's get those nailed down first and then we'll take it from there. Yogesh

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freeswitch Scenario
by Yogesh Patel 17 Mar '08

17 Mar '08

HI Here what we want to do with the switch. Our scenario is like few wholesale carrier will sendus traffic for different destinations, which we will be sending to thedifferent vendors. So we were using 206.222.8.58 as our live switch to testfreeswitch. So we need to understand howfreeswitch is terming following things and where do it makes them so we cankeep adding more customers and vendors. Once it starts working for us. Scenario 1 customer name “Allianz” will be sending call to us from IP 209.62.77.114. presently we can configure this customer as a SIP but we also want to do h323 there. (we don’t authenticate any customer for user/pass, they only come with IP and prefix) they will send a call to us for destinations like India(91) and Egypt(20) so we have to create appropriate dialplan for them. for India(91) calls should go to indiavendor1 on ip address 122.252.236.10 as a 1st preference and once that is fill it sld go to indiavendor2 on IP address 122.252.236.11 calls for Egypt proper (20) and Egypt Mobile(201) should go to Egyptvendor1. Scenario 2 one SIP user created on freeswiotch which we can use to make test calls to the Gateways(vendors) we add to the switch for the testing purpose. that SIP user should be able to call any existing vendors/gateways for different destinations and also new one we will add in future so I think this one will have different dialplan. Scenario 3 1. where do we have to create SIP users which we can give it to retail customersfor them to cone in any ip, means they have to have user/pass and ipprovided by us to let them register in our system and once done they can make acall out NOTE: where it will dump CDR for all above calls ? Thanks Yogesh.

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TEST TO -dev LIST, IGNORE
by Michael Collins 17 Mar '08

17 Mar '08

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