similar to: MeetMe Hints

Hi I've a queue which has generic zap extensions (of my legacy PBX which is connected to asterisk via cross over on span 4 ) logged in ..The legacy pbx extensions are able to logon to queue perfect.. but Whenever a call comes in queue the status of that extension in "queue show <queuename>" always shows as "NOT IN USE" instead of ringing or In use as shown in a SIP

Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). ***** A few details on the box: cat /etc/redhat-release CentOS release 5.10 (Final) arch i686 uname -a Linux hermes 2.6.18-371.4.1.el5 #1 SMP Thu Jan 30 06:09:24 EST 2014 i686 athlon i386 GNU/Linux asterisk -r Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium,

The Asterisk Development Team has announced the release of Asterisk 1.4.35. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.35 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.4.35. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.35 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

I've got a context in my dialplan like so but pressing 0 doesn't seem to be working. Instead of dropping out to the "o" extension, it's just returning to the start of the direcotry app. Same with star. Anyone see where I've gone awry? [attendant] ; <snip> exten => *,1,NoOp(Attendant: Directory) exten => *,n,Directory(default,attendant,eb)

New installation from AsteriskNow 3.0.0 with asterisk 11 and freepbx, running Digium D40 and D70 phones. Direct-dialed extensions work fine, but extensions in RingGroup won't ring - dialparties.agi apparently removes them from the dialstring pre-emptively: dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN) dialparties.agi: Extension 2010 has ExtensionState: 4 What can I do? Any extension