similar to: changing "Nobody picked up in 30000 m"

how to edit the time "30000 ms" for ringing to "40000 ms", i ve tried but i dindt know how,so please help me please. __________________________________ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html

i dont how to edit the time for ringing "30000ms" to "40000ms" when it displayed on console "Nobody picked up in 30000 ms" and its very short time for ringing . please if anyone can help me do it please. ____________________________________________________ Sell on Yahoo! Auctions ? no fees. Bid on great items. http://auctions.yahoo.com/

what is the most stable linux that we can build business on it, i mean the best linux a linux without problems . __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

in routes pattern i tried to write pattern to usa destination and that was 1* it worked well but when i wanted to specify the number of digits then i tried 1NXXNXXXXXX but i didnt work.so i dont know what to write please help. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

i tried to write to usa destination 1* it worked well but when i tried to specify the number of digits i wrote 1NXXNXXXXXX but it did'nt work.can anybody help me please please. ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

Hi: Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are being installed to /lib/modules/2.6.8.1-12mdkcustom how can i fix this up, any one have an idea? Best Regards; Wissam _________________________________________________________________ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://www.toolbar.live.com

ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar – get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. Best Regards; Wassim _________________________________________________________________ Windows Live Spaces is here! It?s easy to create your own personal Web site. http://spaces.live.com/signup.aspx

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I'm using the Local channel in an app of mine and I'm finding that the app is being cut out of the call path. You used to be able to avoid this using the \n command but that doesn't seem to work any more. This is on a recent version of Asterisk. Any comments/suggestion? Darren Wiebe darren@aleph-com.net -----BEGIN PGP

hi: In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i added polarity reversal property but for fxo number 4 i didnt add polarity reversal property but it still giving me on cosole that fxo number 4 is polarized (because the line on fxo number 4 is not polarized). what i want to do is to not let polarity reversal take effect on fxo number 4. that what i have in my

Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with "no, this is absolutely not for doing call marketing". I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students

Hi, I have a client who has asked me to look into the delivery of 30 second audio messages to a list of opt-in customers. Probably looking at about 5,000 messages a week over a 6 week period. I know that this would be a piece of cake to have someone develop but I thought I would ask here first if someone is already doing this and what they would charge to take this on as a hosted solution

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

Hello, I would like apply different call rate (tarif) per outgoing number (or group of phones, based on prefixes), I'm playing with astpp, but seems, that this feature isn't available here, can you recommend any other open-source billing (A2billing, AstBill?), that this can do? thank you! PJ

Hi, What do I have to do to get local\number to work in a context? It works from my [from-internal]... however from subcontexts it does not work: Jan 6 15:55:32 VERBOSE[20237] logger.c: -- AGI Script Executing Application: (Dial) Options: (Local/570323xxxx) Jan 6 15:55:32 NOTICE[20237] chan_local.c: No such extension/context 570323xxxx@default creating local channel Jan 6 15:55:32

I would like to know if any body tried to connect gsm gateway with polarity reversal to fxo module at asterisk server ,and if the polarity reversal solve the problem of the answer and hangup supervison on calls .i appreciate any help.Thanks in advance;Wassim _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's

Hi all: when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. _________________________________________________________________ Windows Live?: E-mail. Chat. Share. Get more ways to connect.

i ve downloaded asterisk-oh323-0.6.6.tar.gz I am getting this and anybody know howto fix this? #tar zxvf asterisk-oh323-0.6.6.tar.gz oh323]# cd asterisk-oh323-0.6.6 asterisk-oh323-0.6.6]# ls asterisk-driver CONFIGURATION Makefile rpm TESTS BUGS COPYING README rules.mak wrapper asterisk-oh323-0.6.6]# make for x in wrapper asterisk-driver; do make -C $x

I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by root@............... on a i686 running Linux. I just spent some time in testing this. I tested the local and IAX2 trunks. Both worked flawlessly. Any comments? Darren Wiebe darren@aleph-com.net