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...; callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line => aaln/3 callerid= "test" <8000104> callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line => aaln/4 extensions.conf [MGCP] include => Toll Free include => CreoLink exten => 8000100,1,Dial(MGCP/aaln/1@192.168.11.200,30,rt) exten => 8000100,2,Hangup exten => 8000101,1,Dial(MGCP/aaln/2@192.168.11.200,30,rt) exten => 8000101,2,Hangup exten => 8000102,1,Dial(MGCP/aaln/3@192.168.11.200,30,rt) exten => 8000102,2,Hangup exten => 8000103,1,Dial(MG...

Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/8cdbeeec/attachment.htm

Hello, This is the scenario i want to setup: Cisco ATA 186 -----------> SER -------------> Asterisk I want the Cisco ATA to register to Asterisk through SER. when the Cisco ATA place a call, SER querry a data base (MySQL or else), and if there is a valid Account for the ATA, the call go to Asterisk. Did someone know how to set SER to work like this with Asterisk? which version of SER

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Hi list, I have successfully upgrade my Asterisk V1-0-RC2 to V1-0-5, but I have a problem. The Asterisk box crashes now every time. I'm using asterisk-oh323. is there a stable version of asterisk-oh323 that can work with the v1-0-5 of Asterisk. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Hello all, Can anyone help me out with this issue ?? I got ASTCC running, but the audios doesn't match my needs (currency, etc.). is there any way to create my own audios and replace the current one?? Thanks. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Derek, I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ?? Thanks. Daniel. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Conniffe Sent: lundi 7 f?vrier 2005 11:59 To: Asterisk

Hello I got the similar error while trying a call. -- Executing Answer("SIP/8000104-86ef", "") in new stack -- Executing Wait("SIP/8000104-86ef", "2") in new stack -- Executing AGI("SIP/8000104-86ef", "areskicc.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php areskicc.php:

Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing "asterisk -vvvc". Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on

Hello, I have some problem using ASTCC application. I've installed the application and everything works well. I've created card numbers, routes trunk and others. When I dial the desired number (77) in my case, I'm prompted to enter my card number. All goes well till I'm prompted to enter the destination number. When I enter a destination number, the system says it's not a