I am using a mix of Call files and AMI telnet from a perl app to place
calls.I sometimes get this in t..
Ive been tasked with finding and implementing a CDR/Queue analyzer to provide information to management about the call centers performance.My Google-fu seems to be returning a lot of things that are more or less abandoned projects.Does anyone have ..
The AstLinux team is happy to announce the release of version 1.0.2.This release features several security updates.All current users are encouraged to upgrade as soon as possible.Please see the documentation at http://doc.astlinux.org for upgrade deta..
We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are: ERROR res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks WARNING app_meetme..
Hi. I am new to asterisk. I have an ivr application with asterisk and voiceglue. I make a call from asterisk (say to A) and when callee press a button voiceglue transfer the callee to another number (say to B). When I look cdr records the billsec betw..
All, I dont know its possible or not. I want to do run time codec selection by asterisk. I have an account in sip.conf and I have active only g729 on it  ……. disallow=all allow=g729 ……. When try to dial that number when I want to use co..