similar to: UserEvent() with multiple body lines

Displaying 20 results from an estimated 700 matches similar to: "UserEvent() with multiple body lines"

2009 Apr 23
0
UserEvent doc : is Uniqueid mandatory in 1.6
Hello, I'm using 1.6.1-rc4. When sending a userevent, (with "UserEvent(MyEvent);" in extensions.ael), I've got : Event: UserEvent Privilege: user,all UserEvent: MyEvent I can't see any Uniqueid field as mentioned http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent or http://www.the-asterisk-book.com/unstable/applikationen-userevent.html Is this Uniqueid mandatory ?
2009 Apr 24
1
FOP and UserEvent()
Hi all, I try to install FOP. It's very nice. In documentation I red that from my dial plan I can launch a popup window with UserEvent() application. I try to follow FOP documentation but I can't popup anything. My structure is: - server 1: Asterisk system - server 2: FOP system - client On client I connect to FOP panel, but I don't see any popup. Someone can help me to configure FOP
2005 Sep 30
4
G.729 patent in France
Hi all, I am building an Asterisk PBX with voicemail and music on hold functions. An ISDN BRI line will also be available and G.729 IP-phones will be used. Are there patents rights applicable to France? Which licence could I use and how many ones are required (only one per phone or also for voicemail and MOH)? Regards Amaury -------------- next part -------------- An HTML
2006 Mar 30
1
internals and ISDN calls fail when Internet is down
Hello, I have an * server configured with a voip provider and an ISDN BRI backup line. When Internet is down, internal SIP calls or ISDN calls fails. I have analysed some internal sip traffic and asterisk doesn't answer fast enough to phones. It sends DNS queries to find voip provider IP but as Internet is down, results are never returned. I have tried to upgrade from * 1.0.9 to 1.2.5
2006 Oct 18
1
[PATCH] Compiz Events
I wrote this quick patch because I want plugins to be able to communicate with each other with events. Using the option values to communicate with each other is a bit cumbersome if you want to monitor for changes or do anything which requires events. It is just a small patch and works in the compiz way (ie by wrapping the core event). Some potential events that I can think of at the moment are
2011 Jan 03
8
Heroku, needs constant AppController updates?
Dear All, Fairly new to rails and Heroku, so could be doing something wacky - do let me know if you think my code practice is off, even if unrelated to this error, I''d like to learn! I''m using Rails 3.0.0, ruby 1.8.7, and ''sqlite3-ruby'', ''1.2.5'', :require => ''sqlite3''. I''ve got an application that goes off and
2005 Sep 14
0
Dial Application Return Codes - Help needed
Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
2005 Sep 15
3
${DIALSTATUS} problems
Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
2003 Apr 16
2
barplot2
Hello, I get a nice looking barplot using the barplot2 function in the gregmisc package: body2 <- barplot2(hh3, beside = TRUE, col = c("mistyrose", "lightcyan"), .... cex.names = 1.0, plot.ci = TRUE, ci.l = cil, ci.u = ciu, plot.grid = TRUE) box() However, obviously I lose the collors when converting from ps to a pdf (outside of R)
2006 Jun 19
0
asttapi 0.10
Hi, I have been playing around with the latest release of asttapi and I have found the 'hangup' problem already reported to the list here <http://lists.digium.com/pipermail/asterisk-users/2006-May/151260.html> Apparently hangup should be done by making use of UserEvent commands. So I have configured this context for being used when making calls from outlook: [outlook] exten =>
2009 Apr 22
1
Should you use UserEvents for monitoring calls ?
Hi, I need to monitor call activity from a custom application software. The goal is to display things like who is on call or not, who has forwarded his call to his voicemail, etc ... When using manager's login command with Event parameter set to on, I'm getting tens of events I don't care about but I suppose I won't miss things like transfers, pickups, parking ... Would it be a
2009 Apr 24
1
Hangup Detection After Originate (Asterisk Manager API)
I have written an asterisk manager client which creates an outbound call using Asterisk manager API's Originate action. when the call is connected I run 3 applications on it. 1)read a dtmf digit from user 2)A customized application which I have written,(It plays something to user) 3)Hangup If user hangs up while app 2(see above) is executing I get a 'Event Hangup' from asterisk in my
2008 Apr 24
1
Full queue issues
Hello everyone. I got a little problem in here: I want to set up a queue so that if anything of these happens: a) No agents logged in b) All agents busy Then the user gets diverted somewhere. I used this (for testing purposes only, of course): exten => 7080,1,Answer() exten => 7080,n,Queue(teste) exten => 7080,n,Goto(${QUEUESTATUS}) exten => 7080,n(ERROR),NoOp(${QUEUESTATUS}) exten
2006 Jun 09
1
hangup extension
I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as
2011 Mar 08
1
(fast) AGI and AMI synchronization ?
Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:3333/fetch,queuename) The idea is
2009 Feb 26
3
Question about Do Not Disturb
Hello, Some of my users have phones lacking a DND button. I need to provide an extension they can dial that will put them in DND, i.e. tell the server not to send them any calls until they get off the DND. I've researched it for almost 3 days now and tried a range of configurations. I'm hoping somebody here has an answer. Currently, I have this in extensions.conf [app-dnd-on]
2009 Jul 07
0
[PATCH server] Update app to work with rails 2.3.2
Note that this does not fix gettext for app, that will be done separately in another patch as F10/F11 require different setups for that. In the meantime gettext works if manually changed in environment.rb to gettext_rails instead of gettext/rails Signed-off-by: Jason Guiditta <jason.guiditt at gmail.com> --- src/app/controllers/application.rb | 200 --------
2009 Jan 28
1
Scope of variable
I have this extension: exten => 1322,1,Answer() exten => 1322,n,Set(CfMC_AMDValue="NotChecked") exten => 1322,n,GotoIf($["${CfMC_DoAMD}" != "Yes"]?NOAMD) exten => 1322,n,AMD() exten => 1322,n,Set(CfMC_AMDValue = ${AMDSTATUS}) exten => 1322,n(NOAMD),Wait(1) exten => 1322,n,UserEvent(E1322-1,${CfMC_ActionID}=${CHANNEL} & ${CfMC_AgentToUse}
2014 Jan 13
0
How to get ringing sound in outbound call in asterisk
I have two server Server_A(outbound call) for agent login and agent make a outbound call from here and pass into server Server_B call extension.conf exten => _91XX.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR) exten => _91XX.,n,hangup() Server_B[192.168.53.197] for call forwarding extension.conf exten =>
2009 Jun 26
0
Problem with RetryDial
I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 10 seconds Asterisk exits the RetryDial step with HANGUPCAUSE=0 and