similar to: Musiconhold instead ringing

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

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

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

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

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

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 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

25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: >> I want to change call files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid

Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road,

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

Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far: [incoming-fax] exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM}) exten => s,2,ReceiveFAX(${FAXFILE}.tif) exten => s,3,Hangup() exten=>h,1,System(/usr/local/bin/fax2mail --cid-number "0${CALLERIDNUM}" --cid-name "home fax"

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 All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers Steve

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/

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!

Is there a variable for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next