similar to: knopsterisk

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

i will be installing asterisk; before that i understood i need RHEL3 or CentOS and i downloaded asterisk@home1.0 already int this download page http://sourceforge.net/project/showfiles.php?group_id=123387 i have seen Download asteriskathome-1.0.iso and Download asteriskathome-1.0-md5sum.txt please anyone explain __________________________________________________ Do You Yahoo!? Tired of

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US

Hi list, I'm using asterisk to send alerts via phone or fax, by using the spool functionnality (writing files in /var/spool/asterisk/outgoing). The system works, but sometimes a file in the outgoing directory is simply ignored by asterisk (nothing on the cli); the spool file is correct. It happenned only twice since the system is installed (about two months), but since it is an alert

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take

Has anyone successfully used a standard bluetooth enabled system to connect to a standard bluetooth enabled mobile phone (not the bluetooth to FXS converters) to create an audio path for phone calls with asterisk, if so is there a writeup on what was done so that others can replicate this. What I am thinking is that via alsa/oss/whatever you should be able to use the bluetooth audio channel as a

Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others.

Does anyone have a mirror of this running? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050715/5fe78cd2/attachment.htm

There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of

Guys, is Steve's ftp site down? DNS says ftp.soft-switch.org doesn't exist. Anybody else seen this?

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

Or you can use authenticate() and have it take its 'passwords' form a text file on your machine. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > trixter aka Bret McDanel > Sent: Sunday, January 29, 2006 5:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion

Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a

has anyone else noticed what appears to be a threading issue in asterisk 1.2.9 (it broke sometime between 1.2.4 and 1.2.9) where if you have about > 50 calls and do asterisk -rx show channels it will display the header but nothing about channels, total calls, active calls, etc. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402

Anybody doing email to fax using spandsp?

Thought those that use cisco in conjunction with asterisk may want to read this. I dont use cisco so I havent read it to see if its actually anything new. Vulnerabilities in Cisco's VOIP system http://www.computerworld.com/securitytopics/security/story/0,10801,103240,00.html?source=x73 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402

Hi All- I am trying to create a post call survey application. I would like to: 1. ask the caller if they want to take a survey after their call completes 2. If no, just transfer the call 3. if yes, 4. bridge up another extension 5. wait for that extension to hang-up 6. have the system (not the user) transfer the call to different extension that administers an IVR based survey. Anyone

So it looks like Livevoip went Bankrupt ------------------------------------------- There is a Federal Court Order in place and has been since Friday early a.m. ALL Suppliers are now under a Court Order that prevents them from terminating any and all services to LiveVoip LLC. If they take such any action they will be in direct violation of a U.S. Federal Court Order. If you have any questions

this is not just playback recorded voice. this is let asterisk say chinese. how to do this. there have any ideas? -- Jeffery iaxtel Num: 1-700-576-1311 fwdnet Num: 728150