similar to: automatically start application from thecommandprompt

Hello, I'm using X-web lite in a webpage to connect to one of our asterisk server. But now I have a problem, when you are connected to a voice script the voice will not be heard after a couple of seconds. When you press or say something that the voice will come back for a couple of seconds. When I thy X-Lite (stand-alone version) I had the same problem, but when I turned off the

Hello, To connect to our oracle database from an asterisk application we use res_perl. Sometimes one of our asterisk server will 'freeze' and work anymore. I have to kill the job safe_asterisk and start it again, so that the application asterisk works again. If I look in the log files it look like that asterisk will 'hang or freeze', if two callers calls exactly at the

Hi, I'm using frequently the perl api within asterisk. Now I'm looking for documentation for the perl commands. Some perl commands I found on this URL: http://www.voip-info.org/wiki/view/Asterisk+PHP Does anybody got more documentation or where I can found some more documentation about perl commands Kind Regards. Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

Hello, I want to record the message of both parties during a dial. My extensions.conf at the line where dial is looks like this: exten => s,803,Dial(SIP/arjankroon2,30,rR) My Sip.conf look like this: [arjankroon2] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend

Hello, I'm trying to get the following situation. Someone calls an application on one of our asterisk server. In this application the caller will call a SIP client. (with the command Dial) The Sip client is connected with another asterisk server. (see below) Caller --> asterisk01 (incoming server) --> asterisk00 (outbound server) --> SIP client (X-lite) Do anybody now how

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

Hi, How do you all handle the situation when a centrale fastagi server process(es) are down? AGI(..) prints "Unable to locate host" and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? thx Arjan Kroon Mobillion BV

Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read

Hi, We are using asterisk version 1.0.0. For queue'ing we use the rawplayer script to play a music file in the background. Now we see that after a while all the sessions on our Linux environment will be taken by the rawplayer process. An example of such a session is (done with ps -ax|grep rawplayer) 24785 ? Z 0:00 [rawplayer <defunct>] 8415 ? Z

Hi, Is it possible to use a live audio stream in asterisk I want to call a number and then hear an external audio stream. For example http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx I thought it was possible to use musiconhold, but I did not get it working. This is my musiconhold.conf ; ; Music on Hold -- Sample Configuration ; [general] [default] mode=custum

Hi, The ami manager call out with an originate through dadhi to a local number (A). If this call is answered, then the ami manager redirect this call to a dial command. This dial command calls through dadhi to another local number (B). Number B answers this call and number A en B are connected. If number B and number A hangs up, there is will be no CDR be written If the dial command is commented

Hi, Does asterisk version 1.6.2.12 support COLP (COnnected Line Presentation) Regards Arjan Kroon Mobillion BV -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110330/937ec861/attachment.htm>

I echo the sentiment that you should just run Asterisk on some small hardware....in an appliance like fashion. In fact, just yesterday I posted an overview of hardware suitable for DIY appliances. I've used many of the platforms mentioned. http://www.mjgraves.com/2010/07/08/d-i-y-asterisk-appliances-a-question-of-scale/ Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262

Hi all, Many thanks for your replies! Will tell my friend and see what he will be interested in. Many thanks! Christian -----Ursprungligt meddelande----- Fr?n: mgraves at mstvp.com Till: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Skickat: 10-07-09 15:29 ?mne: Re: [asterisk-users] Pbx_f?r_Windows?_-_Email_f ound_ in_subject I

Hai, Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the Connected Line ID, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV

It may have gone to sleep. Chris Cooper Systems/Network Administrator EFC International 1940 Craigshire Blvd St. Louis, MO 63146 US Phone - 314-439-4325 Fax - 314-439-4443 Mobile - 314-402-8912 - -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent:

Hi, I'm using asterisk version 1.8.3.3. In earlier versions I used queues, but with the new version the queuing mechanism doesn't work If I look in the CLI at I see that the queue-member is invalid: Members: DADHI/g3/0655871460 (Invalid) has taken no calls yet The queues.conf looks like this: [general] persistentmembers = yes monitor-type = MixMonitor [test] musicclass

The PostgreSQL Conference East is in NYC this year and the schedule has just been published. As a note there is a full day RoR class happening at the conference. March 08, 2011: Celebrating 15 years of PostgreSQL, early. Following on the smashing success of PostgreSQL Conference West, The PostgreSQL Conference for Developers, End Users and Decision Makers, is being held at the Hotel

Hello, I'm a newbie in connection several asterisk servers with each others. I've got the following situation. I've got 9 asterisk servers (asterisk00 till asterisk08). When I call to asterisk08 then I want to redirect an application which runs on asterisk00. But how can I redirect in an application on asterisk08 to an application on asterisk00? Or isn't this possible?