search for: terracom

Displaying 9 results from an estimated 9 matches for "terracom".

2004 Sep 27
1
ASTCC installation problems
...ysql" at /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi/DBI.pm line 744. Perhaps the capitalisation of DBD 'mysql' isn't right. at ./astcc-admin.cgi line 43 make: *** [install] Error 255 can anyone help me with that? Thanks Habiyakare Aimable Voice Services Terracom Communications Tel :(250)08435550 SIP:04400104@voice.terracom.rw E-mail:aimable@terracom.rw MSN:aimable@terracom.rw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040927/a8945cba/attachment....
2004 Oct 06
1
Anyone using VoiceMaster
Is there anyone with experience how to integrate Sysmaster's VoiceMaster? Please can you share your experience. Thanks. Habiyakare Aimable Voice Services Terracom Communications Tel :(250)08435550 SIP:04400104@voice.terracom.rw E-mail:aimable@terracom.rw MSN:aimable@terracom.rw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041006/50419015/attachment....
2004 Jun 21
4
disabling ALERTING message
Hi all, Is there a way of disabling ALERTING message on a PRI channel? I have a problem .* is sending ALERTING message to the Nortel telco switch of my local provider BEFORE it dial the number it has to .if the number is busy or invalid there is no way we can tell this to the switch because it has already been told that the phone is ringing I am using asterisk Asterisk CVS-04/06/04-10:46:21 with
2004 May 05
1
Problem in Extension.conf
Hi, Have a problem in my extension.conf: I have: [sip] exten => _333.,1,wait,3 exten => _333.,2,Answer exten => _333.,3,AbsoluteTimeout,7 exten => _333.,4,Hangup I wanted to test if * is executing this dial plan by calling 3335254255 for example. The problem is as follow: It waits, it answers but it does not seems to see the Absolutetimeout: call goes forever. What's wrong? Am
2004 Jun 07
0
FW: Problem with Asterisk PRI forwarding to SER
_____ From: Habiyakare Aimable [mailto:aimable@terracom.rw] Sent: Monday, June 07, 2004 11:49 AM To: 'asterisk-users@list.digium.com'; 'gt'; 'support@digium.com' Subject: Problem with Asterisk PRI forwarding to SER Hi all, I have a problem. We have a phone system setup like this: SIP phones------------>SER-------------...
2004 Jun 17
4
Problems with PRI with T410 messages
...phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Phone Services TERRACOM Broadband aimable@terracom.rw -----Original Message----- From: asterisk-users-request@lists.digium.com [mailto:asterisk-users-request@lists.digium.com] Sent: Thursday, June 17, 2004 10:56 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs Send Asteris...
2004 May 05
0
Problem with extension.conf
Hi, I am new on this field and I'm looking for some help I have this conf: [sip] exten => _333.,1,wait,3 exten => _333.,2,Answer exten => _333.,3,AbsoluteTimeout,7 exten => _333.,4,Hangup this works up to answer. then it can not stop after the 7 sec i specified in AbsoluteTimeout. Why? Any help? Regards Ghislain
2004 May 06
0
Problem in extensions.conf
Ok I tried but it does not work: now the settings are as follow exten => _123.,1,Answer exten => _123.,2,AGI(test.agi) exten => T,1,hangup the AbsoluteTimeout(5) is in test.agi (PHP) I put "AbsoluteTimeout" before "Answer": when i call for e.g 123456 it tries upto timing out. So I put again "Answer" and then "AbsoluteTimeout" then the last AGI
2004 Jun 11
2
Asterisk PRI calls to SER problem
Hi all, I need help. I have a Linux box with SER as a proxy server with ip phones attached on it , and another linux box with Asterisk and T410 card connect to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk and then to SER box and then to the phone .every time an invalid number dialed from PSTN to SIP phones connected to SER asterisk says that the call is progressing