Re: [Freeswitch-users] sip message logging and analysis

i agree with christian, though i would use tshark. you can actually get the fields you want (method and callid) and store them in a dB. then you need to match them with a query. it is simple but Lots of work. look into -e and -E of tshark separate the fields by "," have fun! David El 18/12/

Re: [Freeswitch-users] FS & Wikipbx Help

Hello Fernando, The best you can use is this one: http://wiki.freeswitch.org/wiki/Getting_Started_Guide very easy ;) David On Sun, May 24, 2009 at 5:35 AM, FERNANDO VILLARROEL wrote: > > Dear All, > > I am new user of Freeswitch and searching information i find wikipbx: > > http://www.wiki

Re: [Freeswitch-users] Country Codes + Area codes

Those are all paid-services.. :S d On Sun, Aug 31, 2008 at 2:36 PM, sam bofill <[EMAIL PROTECTED]> wrote: > Try http://www.numberingplans.com/ > > --- On Sat, 8/30/08, Seysan <[EMAIL PROTECTED]> wrote: > > > From: Seysan <[EMAIL PROTECTED]> > > Subject: [Freeswitch-users] Country Codes + Area c

Re: [Freeswitch-users] GUI

Oh! And something else, Mera says they have 1000+ developers... well the Opensource community have MILLIONS! On Fri, Aug 1, 2008 at 5:19 PM, David Villasmil < [EMAIL PROTECTED]> wrote: > I would never BUY a Mera, I have lots of experience with Mera, and although > the system is no

Re: [Freeswitch-users] GUI

I would never BUY a Mera, I have lots of experience with Mera, and although the system is not bad at all, IMHO it is way too expensive given all the OpenSource projects out there. AFAIK, FS can hold at least 300+ simultaneous calls. But even if there wasn't a FS, there's Openser and there's Asteris

Re: [Freeswitch-users] GUI

So... anyway to see it working? or is this commercial? thxs d On Thu, Jul 31, 2008 at 10:32 PM, Lito Manansala <[EMAIL PROTECTED] > wrote: > www.wikipbx.com > > 2008/8/1 Ilan Perez <[EMAIL PROTECTED]> > >> Anyone developing a GUI for freeswitch >> >> >> >> >> >> *Ilan Perez* >> >> *Webmaster*

Re: [Freeswitch-users] FreeSWITCH 1.0.1 "Phoenix" Released!

Hello, Seems like nice improvs, is there anywhere I can see the actual ASR and TTS working? Cheers, David On Thu, Jul 24, 2008 at 1:47 PM, Michael Collins <[EMAIL PROTECTED]> wrote: > wikipedia and voip-info.org have been updated. > -MC > > On Thu, Jul 24, 2008 at 10:06 AM, Anthony Mines

Re: [Freeswitch-users] g.729

Thanks! That'll be a blast! hehe! d On Thu, Jun 26, 2008 at 2:08 AM, Brian West <[EMAIL PROTECTED]> wrote: > Not yet. Stay tuned! > > /b > > On Jun 25, 2008, at 6:42 AM, David Villasmil wrote: > > > Hello all, > > > > Is there a licensed g

[Freeswitch-users] g.729

Hello all, Is there a licensed g.729 yet? When will we be able to transcode from another codec to g.729?? I know we can pass-through, but transcode? cheers d ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.fre

Re: [Freeswitch-users] IAX Trunking

t's > 120 ms of audio you just lost - probably quite noticeable. Depending on > the jitter buffer being used, it may also want at least a few packets in > the buffer before it starts playing audio. If 3 x 120ms packets is what > it wants, there will be a 360ms delay at the start of the call

Re: [Freeswitch-users] Success stories

500K? Wow, that's a lot a single machine... I'm already installing! ;) d On Wed, Jun 18, 2008 at 11:23 AM, David Knell <[EMAIL PROTECTED]> wrote: > Hi Jim, > > We passed interconnect testing with BT earlier this year using FS > without problems - > I understand that Nextone and Quintum users ha

Re: [Freeswitch-users] IAX Trunking

Thanks for the reply, I was just curious because we have many gws doing IAX trunking to save bandwidth as the gws are behind ADSL. Thank you again. d On Wed, Jun 18, 2008 at 11:18 AM, David Knell <[EMAIL PROTECTED]> wrote: > Hi David, > > I believe the answer to be 'no' - the workaround

[Freeswitch-users] IAX Trunking

Can FS do IAX Trunking? David ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freesw

Re: [Freeswitch-users] equivalent of realtime

Oops! sorry On Fri, Jun 13, 2008 at 11:08 PM, David Villasmil < [EMAIL PROTECTED]> wrote: > As far as user registered is concerned: YES, take a look at usrloc module > with mysql > For numbers dialed, look at lcr > > On Fri, Jun 13, 2008 at 11:05 PM, Jonathan K. Cre

Re: [Freeswitch-users] equivalent of realtime

As far as user registered is concerned: YES, take a look at usrloc module with mysql For numbers dialed, look at lcr On Fri, Jun 13, 2008 at 11:05 PM, Jonathan K. Creasy <[EMAIL PROTECTED]> wrote: > Can I load SIP peers and extensions from a MYSQL database with > Freeswitch? > > > > -Jonathan >

Re: [Freeswitch-users] Startup pointers

Mike, Thank you for answering! Actually I went into the irc yesterday and got lots of help. It so happens that conf files were not put where they should've been by the installer (somehow). I did a make sync (I had the sources from two days ago) and re-did everything and its all A-OK now! I go

[Freeswitch-users] Startup pointers

Hello all, I just downloaded FS from svn, compiled, everything went smooth. (all default) I can actually register 2 UAs, 1000 and 1001. I can call the conference, i can call 888, etc. BUT, I can NOT call the UAs! The question: How does FS actually work? I mean, how does SIP UAs area mapped