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Hi i installed ceptral and i want to test it with asterisk can u plz tell me if i was wrong here>> ?? exten => 2,1,Answer exten => 2,2,system(/opt/swift/bin/swift "hello world") exten=> 2,3,Hangup() Mahmoud Badran ATSI Tel: +20 2 607 8917 Fax: +20 2 607 9178 gsm: +20 105674409 mahmoud.badran@atsint.com -------------- next...

Hello ; how can i enable billing only while using specific trunk (ex:zap) but internal sip calls will not be counted specifically how to make all outbound is counted i am using asterisk mysql cdr enabled -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050626/0faf0974/attachment.htm

AVE! i am trying to register h323 asterisk to the gatekeeper as i installed asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323 on fedora core3 on a cisco mcs 7800 server problem is i want the asterisk to register with gatekeeper endpoint with specific zone name and type... i searched the web, mail list but there weren't any helpful ones could anyone plz tell me how to

try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To:

Hello, I'm looking to connect Asterisk with three (four in the future) PSTN lines, and would like to get some opinions on the TDM400 Digium card, vs. sip gateways like the Mediatrix 1204, vs. other hardware solutions I'm not yet aware of. I need the ability to prioritize which PSTN lines are used for outgoing calls (I understand this can be done with the Mediatrix --

hello u can see the readme.udev in the zaptel directory that's normally answers ur question ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Ian Bert Tusil Sent: Fri 7/1/2005 9:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got this error after my installation when i doztcfg -vv how can i solve the error on the last

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM