search for: 20asterisk

...made a simple menu using the Background application and some wav files. I converted the wav files using for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm"; done (from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk) The first two files "01/bemvindo" and "01/menu_top" are good. But the third file (01/menu_top), fails in the end of the sentence, and this message "Auto fallthrough, channel 'SIP/dov.bigio-ae4a' status is 'UNKNOWN'" appears in the console. -- Exec...

Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no

...call center environment. A few considerations: 1) USB headsets are preferable, because they don't require a soundcard. 2) Omnidirectional microphones are problematic, because they pick up too much background noise. Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

...ch as the TE410P. Any answers would be greatly appreciated. 1) Do they support standard T1s or are they ISDN-only? 2) Do you know of anyone offering support for configuring T1s for Digium cards, and if so at what cost? Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

...alled. It is preferable for the server to have a single CPU and no shared IRQs. I would really appreciate any comments regarding the feasibility of this scenario, as well as any suggestions about hardware requirements. Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

...tudio for free, and you can access it through SIP, so I thought this would be a fun and cheap way to integrate voice recognition into my IVR. I have posted a brief tutorial with code and examples on the voip-info.org wiki ( http://www.voip-info.org/wiki/index.php?page=Add%20Voice%20Recognition%20to%20Asterisk ) as well as at my blog ( http://www.spinepunch.com ). The code is a little rough, and I'm sure there is a better way of doing what I did, but this was easy and it worked for me. What's next on my to-do list is trying to cover up the TellMe jingle before it starts the VoiceXML app. If anyo...

Hello, I'm using Asterisk 1.6.2.0-beta3 with asterisk-addons-1.6.2.0-rc1 to write cdr into mysql. I followed: http://www.spiration.co.uk/post/1327/asterisk-addons%20setting%20up%20mysql%20cdr%20for%20Asterisk, and it is mostly working however CLID is not written to mysql, although it appears in Master.csv. What do I possibly missing? TIA -- Best regards, pepesz76 mailto:pepesz76 at o2.pl

Hi all, I put together a couple of new tutorials on compiling Asterisk 12 with PJSIP on CentOS 6.5 and test-driving ARI on the same box. You can find them at: http://astrecipes.net/index.php?q=AstRecipes/Compiling%20Asterisk%2012%20on%20CentOS%206.5 and http://astrecipes.net/index.php?q=AstRecipes/Getting%20started%20with%20ARI Comments welcome and happy holidays! :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com

1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545

How to enable cdr_mysql.conf in Asterisk 1.6? I have installed asterisk-addons which compiled mysql support, "module show" is showing "cdr_addon_mysql.so" but cdr_mysql.conf was not created in /asterisk directory Is there any configuration file to enable mysql support? Comping cdr_mysql.conf from previous installation does not do anything, calls aren't recorded. --

i have got a music file with extension mp3 and it is not workign with background() is there any way to convert the mp3 to gsm or any other codec? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050222/2ba5a4f0/attachment.htm

Hello list: I have a very odd problem. Seemingly randomly, my Polycom IP600 phones will ring without a call being placed to it. That is to say, a random phone will ring. Nothing shows up under Caller ID. Even the buttons that light up to show an incoming call do not light up. If you pick up the handset, you can hear the phone ring through the speaker. Hanging up the phone makes it stop

Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the

Hi Since Im using the mISDN drivers and no zaptel stuff, I had to install ztdummy to get MeetMe to work. Well, that was the plan. Now, after getting the latest zaptel version over CVS (Im using Kernel 2.6), uncommenting all the modules except ztdummy in zaptel.sysconfig file and compiling this by "make", "make install" and "make linux26", I rebooted and

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem