similar to: asterisk directory dialing

Dear all I have one more question about codec what codec i use for LAN setup G.729 or Alaw which is best for LAN setup caz some people told me G.729 is use for wan link not for lan caz it is cost effective so can anyone suggest me best codec for asterisk and SIP phone Rgds satish patel --------------------------------- Don't pick lemons. See all the new 2007 cars at

Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel --------------------------------- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. -------------- next part -------------- An HTML attachment was

how I can install it on Asterisk so I can use it? Anyhelp? Regards Bilal ____________________________________________________________________________________ Don't let your dream ride pass you by. Make it a reality with Yahoo! Autos. http://autos.yahoo.com/index.html

--- Ralph Giles <giles@xiph.org> wrote: > On Fri, Sep 14, 2007 at 02:51:34PM -0700, Josh Coalson wrote: > > > checked in to CVS is what will be very close to the 1.2.1 release > > of flac scheduled for monday. if anyone can try building it and > > even better running the test suite, and reporting back any > problems, > > that will help me get things in

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Dear all I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3

Dear all I have asterisk 1.2 and now i want to install E1 card with support Q.SIG singaling so which E1 card is best for my setup i need single port E1/PRI card which support Q.SIG Regards Satish patel --------------------------------- Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. -------------- next

Hi Anton, in a solution without CAZ, isNullValue can just return true when it sees the special "this ConstantArray is all zero" flag. So all the places that now look for CAZ can just use isNullValue instead and there need be no performance loss. That said, CAZ is more "in your force" so less likely to be forgotten about. Another interesting possibility is to handle more than

Dear all I am going to install 2 port pri card on asterisk but i dont know how to incomming call goes in to IVR and how to route call outside base on pattern match means if some one call on mobile phone then use PRI 1 and if call on landline phon call route through pri 2 how to make dission base on pattern number Rgds satish patel

Hi Chris. The main question is how to implement ConstantAggregateXXXXX::getOperand? Should it be empty collection, or it must return the item aggregated? -Stepan. 22.01.2012, 04:43, "Chris Lattner" <clattner at apple.com>: > Yep check out PR1324.  Doing something like this would be a great improvement. > > -Chris > > On Jan 21, 2012, at 9:42 AM, Duncan Sands

Yep check out PR1324. Doing something like this would be a great improvement. -Chris On Jan 21, 2012, at 9:42 AM, Duncan Sands <baldrick at free.fr> wrote: > Hi Anton, in a solution without CAZ, isNullValue can just return true when it > sees the special "this ConstantArray is all zero" flag. So all the places that > now look for CAZ can just use isNullValue instead

Hi Stephan, I've been thinking about this a bit and have some more specific ideas, I'll write up a design and maybe implement it over the next few days. -Chris On Jan 21, 2012, at 11:50 PM, Stepan Dyatkovskiy <STPWORLD at narod.ru> wrote: > Hi Chris. The main question is how to implement ConstantAggregateXXXXX::getOperand? Should it be empty collection, or it must return the

Dear all I have asterisk 1.4.11 on Cent 4.3 i have faceing some problem i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make outgoing call it is working fine but incomming call not handling ...when i call from outside to this line it is rinning but no one call land on my asterisk no debug in asterisk some time it land but most of time not .....

Most likely, yes. They should have a copy of the license agreement, and you can take a look at that and see what they have to say about using them apart from the VirtualIron product. The only open source PV drivers are the ones that James has released - search the list for "GPL PV drivers" and that should turn up some threads related to those. The drivers are still very much in the

Dear all I have TDM808B 8 port FXO it is configure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming

How many times will a typical client call getOperand on your helper when it wraps a 10,000 element ConstantAggregateZero? With CAZ the client is force to think about this case, and often handles it much much more efficiently. CAZ is a time optimization as well as a space optimization. -Chris On Jan 21, 2012, at 12:03 AM, Stepan Dyatkovskiy <STPWORLD at narod.ru> wrote: > Hi Chris.

Hi, Is there an easy option in rsync which enables it to sync the data first and then the symlink pointed to it. For e.g. at destination I have a symlink "new" which points to anything latest at source. The problem is that when the sync happens it updates the "new" link first before the data it is pointing to has arrived. This causes me all sorts of problem. Please