similar to: FXO modules and polarity reverse

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

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: 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

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

i dont know how to edit the the time for ringing "30000ms" to "40000ms",please help me. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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

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

Hi: When i get an incoming call, i want asterisk to make the caller hear music"musiconhold" instead of ringing,Can any body help me with this? Best regards; Wassim _________________________________________________________________ Get the new Windows Live Messenger! http://get.live.com/messenger/overview

Hi: Iam using an asterisk server with astcc ,iam facing a problem with astcc that when the call is hangup sometimes astcc doesnt calculate the call cost and the call time and without writing the call status on cdrs table . I tried to run this command "realtime mysql status" on the asterisk console and that what i've got: [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637

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/

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/

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 got a dumb Western Electric payphone and some homebuilt hardware to control the coin relay which is accessible to Asterisk through the AGI interface. I'd like to be able to set the state of the coin relay to collect at the end of a call if a called party answers. [Hey, I admit this project is being persued just for the fun of it ] Looking through the documentation, there is a way to

hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. _________________________________________________________________ Windows Live?: Keep your life in sync.

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

when i tried to compile asterisk-oh323 i get an error that channel_pvt.h is missing,where i can find and download it and in which directory i must put it. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi: i have a hosted server with asterisk and a2billing as a billing plattform, when i am trying to enter the server remotely by ssh, memory error message displayed: -bash: fork: Cannot allocate memory i have 1GB RAM on the system ,and there is 15 to 25 concurrent calls on the system is'nt 1GB of RAM sufficient for this volume of calls on Asterisk. and when iam using "top"

I'm trying to figure out if I'm the only one with these problems with them. I recently had a few customers that switched to them because of the price, of course that means that they have to use FXO ports, but it is realy cheaper, so customers don't really care. In any case, there are 2 issues that I can't get solved, and they are not interested in helping. 1. When they tell you