search for: daniel_eboa

Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP

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

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 5924 bytes Desc: image002.jpg Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/253a6828/attachment.jpeg

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 5921 bytes Desc: image001.jpg Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/74245e28/attachment.jpeg

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 5870 bytes Desc: image001.jpg Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050128/198114e6/attachment.jpeg

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:

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 5870 bytes Desc: image002.jpg Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050129/e53b0567/attachment.jpeg

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