search for: plyschen

Hello List, I'm using a dialstring like the one below. I want to have three different things happening depending on exit cause. Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000])) These 3 things could happen: 1, Caller hangs up 2, Callee hangs up 3, The 20 seconds is up and call is terminated from Asterisk. Is there a way to separate these 3? Thanks, Best regards, Tobias --------------

If i were to use Asterisk with, say 1000 IAXy's, is there a way to provision them from a central spot? It would be very improper to have to let each end-user do this by him/herself... The manual and documentation for the IAXy is very limited, and i can't find anything about this kind of things... Anyone uses the IAXy in an environment like this and can help me with some answers? Thanks

Hello list, I'm having some problem integrating the SELECT TIMEDIFF() and SELECT DATEDIFF() in my code. The syntax I'm using works without any problems if I run them directly from the MySQL Client, but from the Asterisk Dialplan it just wont work. Is there a limitation in the MySQL() application for the Asterisk dialplan that produces this error? <CODE> context testsql { s =>

Hello, I'm having problems with LIMIT_PLAYAUDIO_CALLEE in the Dial application. I want to play the limit file to both caller and callee at the same time, but it plays the limit file first to the caller and then to the callee. I searched the list and found someone with the same problem back in '06, but couldn't find any solution for the problem :( Anyone knows? Thanks, Best regards,

Hello again list. I have gotten some indications that the penalties for agents in Asterisk might not work as intended. Can I get peoples view on this that uses the penalty function in Asterisk Queues? Thanks in advance Best regards, Tobias -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm looking for a free US DID in either North Carolina or Georgia State. I've browsed the list on voip-users, but couldn't really find anything (maybe I'm blind). Does anyone know a provider? Thanks, Tobias -------------- next part -------------- An HTML attachment was scrubbed... URL:

Date: Mon, 23 Jun 2008 08:00:08 -0400 From: "David Backeberg" <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <3de056a30806230500k7e66185l7bfe473ed398ebf6 at

Hello list, I have this strange problem whenever I try to make an "ael reload" from the Asterisk CLI. The command gives the following result and crashes: voip-1*CLI> ael reload Disconnected from Asterisk server Executing last minute cleanups Asterisk ending (0). root at voip-1:/etc/asterisk# As far as I can see, aelparse can't find any errors in my configuration, following