similar to: detecting ring

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

2013/3/29 Julian Lyndon-Smith <asterisk at dotr.com> > check out the endbeforehexten option in cdr.conf > > this needs to set to "yes" > > Julian > Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the "saving to file" part. Then my issue is I can't update CDR value is hangup exten. Here is a

We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian

I'm trying to write a dialplan that will allow me to "stress" test it. I want to be able to dial an extension, or pretend that the extension is busy or out of order (so that I can see what to do) given the dialplan snippet: [outbound] exten => _X.,1,NoOp(${TEST}) exten => _X.,n,Dial(SIP/${EXTEN}) exten => Busy,1,Busy(2) exten => Busy,n,Hangup() exten =>

I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on

Perhaps I've got the whole concept wrong, but here goes: Using 1.4, when someone from the outside dials my direct line (123456), I want it to call my extension at work (SIP/456), my extension in my home office (vpn connection to corporate lan, SIP/678) and my mobile (654321). So my dialplan is thus: exten => 123456,1,Dial(SIP/456&SIP/678&Zap/G3c/07803654321,30) exten =>

I am using AddQueueMember to add a local channel to a queue. My (simplified) dial plan is [AddMember] exten => 789,1,AddQueueMember(SomeQ|Local/456@Agent) [Queue] exten => 123,1,Queue(SomeQ|nt|||120) exten => 123,2,Hangup() exten => h,1,NoOp(InQ) [Agent] exten => 456,1,Dial(SIP/456) exten => 456,2,Set(SomeVar=SomeValue) exten => 456,3,Hangup exten => h,1,NoOp(InAgent)

I'm trying to build a test suite so that I can run "calls" through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: <snip> ; #1 "answer" a call and play music 000XXX : ring for a random period,

As I have no friends and no life I thought that I would set up my asterisk server with Skype. 1) Paid the $, got the licence, built and installed 2) create a business skype account (called company "foo") 3) created a member of the business called "bar" 4) updated the skype conf file 5) restarted asterisk => skype show settings Skype For Asterisk Settings:

Is there anyway on knowing in the "h" extension if a call has been ended as a result of a transfer ? i.e. 1) A calls B. 2) B transfers A to C. 3) B gets hung up. 4) A talks to C at (3) i need to know if this is a normal hangup (A or B has hung up) or if it is a result of the transfer. Julian.

We are in the process of implementing realtime voicemail. I was wanting to "stress-test" the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup()

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

Julian, I did not understand what you meant by this paragraph. "Now, for the twist. For development and testing, I assigned an ip address and gateway to the bridge. I need to be able for a "non-it" person to install this box without having to set it up at all , so it cannot have an ip address assigned, as it *may* be in use somewhere else on the lan or router." Did your

I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls "popping in and out". Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian.

svn trunk 31497 For the life of me, I can't get this :) I want to be able to catch the situation where the calling party hangs up *before* the call is connected to the called party. My dialplan is thus: macro DialExternal(exten) { Dial(Zap/G3/${exten},120,g,M(connected)); goto DialResult|r${HANGUPCAUSE}|1; Hangup(); }; But the goto dialresult is not executed: Executing

Man, I really feel stupid, but after banging my head on a brick wall for several hours ... I need help! I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten 5707, and I've got an xlite on 5608. When I make a call from an outside line, I dial SIP/5608. The little blinky light on the GXP that's monitoring 5608 goes, well, "blink blink". :) I then press

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message