search for: lyta

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2004 Aug 23
2
Bug#267587: logcheck-database: Additional rule needed for postfix
Package: logcheck-database Version: 1.2.25 Severity: normal postfix/smtpd\[[0-9]+\]: lost connection after (CONNECT|DATA|RCPT|RSET|EHLO|HELO|MAIL) from Please include the above line in the ignore.d/server/postfix file. That catches messages that occur very often on busy Postfix servers. -- System Information: Debian Release: 3.1 APT prefers unstable APT policy: (500, 'unstable')
2011 Nov 11
1
What the variable that return the IP Phone username to use it for AddQueueMember
Hi All; To simplify the the login and logout for the agent, I am looking for the variable that can be used for the AddQueueMember (in the place of the ?????? as following: exten => 100,1,AddQueueMember(CustomerSupport,${????????},1) exten => 100,2,Playback(agent-loginok) exten => 101,1,RemoveQueueMember(CustomerSupport,${??????}) exten => 102,2,Playback(agent-loggedoff) In other
2012 Jun 29
0
IAX Trunk issue. (Dale Noll
.../${EXTEN}) Note: It appears that you are doing it correctly from asterisk-1 towards asterisk-2 exten => _5XXX,1,Dial(${IAXTrunk}/${EXTEN}) Assuming, of course, that the variable IAXTrunk is properly set. Dale -- "The truth speaks for itself. I'm just the messenger." Lyta Alexander - Babylon 5 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120629/4bfc7a58/attachment-0001.htm>
2012 Jun 25
1
IAX Trunk issue.
I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it's supposed to, but it's not the tt-weasels under its
2011 Nov 15
2
Goto Queue, does not work, it should play message or any thing
Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten => 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten => 5631040,1,Playback(WelcomeMessage) exten =>
2011 Sep 21
3
RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4
Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on
2014 Jan 20
1
ISDN Cause Code 47 Errors
...om-pstn switchtype=national signalling=pri_cpe pridialplan=unknown prilocaldialplan=national relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes channel => 73-95 ====================================================== "The truth speaks for itself. I'm just the messenger." Lyta Alexander - Babylon 5 ====================================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140120/b9fa79c1/attachment.html>