- FreeSWITCH-users - FreeSWITCH Mailing Lists

Testimonials
by Brian West 02 May '08

02 May '08

I see this page is blank... I can hope that everyone can do the right thing and catalog their experiences in hopes that it will help others! ;) Here is the wiki page http://wiki.freeswitch.org/wiki/Testimonials Brian West sip:brian@freeswitch.org

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EventSocket for .Net
by Jonas Gauffin 01 May '08

01 May '08

Hello Someone was interested in a EventSocket implementation in .Net. I've added by version to the tree now. It can be found in freeswitch\scripts\contrib\verifier. I've also added it to codeplex, http://www.codeplex.com/eventsocket, will also add a binary there in a few days. I've also started to work on a IVR Interface (that allows you to do IVR's directly in .Net with the help of the library). Give me a few days before you try it. Cheers, Jonas

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IP authentication
by Jed Stafford 01 May '08

01 May '08

This feel's like a very stupid question, but i've scoured for hours through the documents, and samples I can find without finding an answer. I'm assuming I'm missing something very obvious. I'm just trying to have freeswitch accept a call from a static IP address, then forward that call to a provider, essentially a very static SIP proxy. I've tried added my IP in question to the acl.xml file, but no success. All calls are rejected with a 407 Auth required. Thanks for any pointers. I think I will write a paragraph or two on this when I get it working, as it's something I think a lot of people are looking to try. Asterisk will not scale enough, and I don't want to do this via OpenSER/RTPProxy, etc. -Jed

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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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Switching from digest to basic authentication
by Maxim Karp 30 Apr '08

30 Apr '08

Hello, I am using a WM6 mobile client to connect to FS that doesn't seem to like digest authentication. How can I disable digest and enable basic authentication? Thanks, Maxim.

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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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getting started help
by Larry Edelstein 29 Apr '08

29 Apr '08

Hi all - I installed Freeswitch a few weeks ago and was never quite able to connect to it via Gizmo or IPKall. I can't seem to configure the directory correctly and get my Gizmo phone to register; my IPKall phone doesn't seem to get anywhere either. I'm a little fuzzy on the exact error messages I got, and before I go run my head into that wall again I figured I'd post for help. I'm pretty sure that the software failed to find the user...extension...in the FS directory. So I couldn't register the...phone. Or something. I'm new to SIP. My main goal is to enable some dial-in and dial-out services from POTS and wireless phones, but getting SIP phones to work would be a start. Although I'm a hugely experienced software guy, I haven't worked with SIP or telephony before. So I am playing catchup. If someone could give me some tips on which files I need to modify to support this scenario, that would be great: - I have an IPKall number, and when I call it, I get to a Freeswitch service. - I have a Gizmo phone, and when I call it, I get to a Freeswitch service. - I can call out from Freeswitch to a SIP phone. - (lower priority) I can call out from Freeswitch to any phone. Thanks Larry Edelstein Now Answers LLC

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uuid_record overlaps channels
by Jonathan Palley 29 Apr '08

29 Apr '08

Hello all - I'm using uuid_record to record calls. We are encountering a problem where the voice for either channels starts overlapping and not being in sync as it is in the actual call. I'm guessing perhaps I'm missing a settings to make sure the calls are mixed in realtime and recorded as they actually sound on the server? Anyone know? I can provide example recording privately. Thanks - Jonathan

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Multiple FS servers sharing registration data
by Mikael A. Bjerkeland 28 Apr '08

28 Apr '08

Hi, is it possible to have one FS server acting as a registrar and several other FS servers using the same registration data for authenticating calls? If so, what are the options? I saw some notions of ODBC support, but no documentation on the tables which must be created. Is ODBC the approach? Regards, Mikael

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Remote SIP extension deployment
by Chris Chen 27 Apr '08

27 Apr '08

Could any of you share your experience with remote SIP extensions behind another NAT? Basically my setup whould be like this: [ Remote SIP extensions]------>(private IP@) NAT Router B---(public IP @B)-------(Internet Cloud)-----------(public IP @A) NAT Router A---(private IP @A)------(private IP@C)[Freeswitch] I have the domain registered as "sip.mydomain.com" let's see, I am trying to config the remote SIP extension as 1018 using port 5080 hitting the public context, but I got the error on the freeswitch console "user 1018(a)sip.mydomain.com not found" Could you guys guide me to the right direction for remote SIP extensions' deploymen? Your helps are appreciate. Thanks, Chris

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