similar to: Random crash of the machine ? due to Asterisk 11

Displaying 20 results from an estimated 10000 matches similar to: "Random crash of the machine ? due to Asterisk 11"

2013 Mar 29
1
Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
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
2013 Mar 28
2
Asterisk 11 -CDR values changed in hangup handler not saved ?
Hello, I'm using Hanhup Handlers in a testing asterisk 11 system. Within one such handler, I'm setting CDR values. To me, it seems those changed CDR values are not saved in CDR back-end. Can you confirm ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:
2012 Jan 04
4
Speech recognition in asterisk using google voice API
Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:
2009 May 24
2
Can I run two instances of asterisk
Can I run two instances of asterisk sharing a single te412p ? I want to be able to have several asterisk servers (for testing various scenarios) running on one server. I was wondering if these asterisk processes could share a zaptel/dahdi card nicely. Julian
2009 Mar 12
8
UK ISDN-30 and ANI
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
2006 Mar 15
3
Double-ring tone
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
2010 Aug 28
1
Play a number of files to a caller
I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the
2007 May 24
3
meetme sounds
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.
2008 Dec 16
4
RDNIS and asterisk
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
2006 Jun 13
3
Queues and macros and agents
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
2006 Jun 14
2
AddQueueMember and Local channels
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
2009 Feb 12
4
Multiple caller id ...
If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension
2006 May 31
2
AEL2 and CID
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);
2009 Nov 25
6
How many lines do you use.
Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a
2007 Mar 19
3
Cepstral and numbers
Does anyone have any idea on how to force cepstral to convert a number to speech ? I have noticed that sometimes it speaks the number correctly, and at others it doesn't. 1) 787 is pronounced 7-8-7 2) 123 is pronounced one-hundred and twenty-three. 1) is wrong for what i need, 2) is perfect. Is there anyway of forcing numbers to be pronounced as 2) ? I've tried looking at the ssml
2007 Oct 18
8
centos 5 vs OpenSuse 10.3
Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian
2005 Nov 08
3
Agent Call Recording
When recording inbound agent calls, if the queues use agent members (Agent/6000), we can get the calls recorded as agent-xxxx.yyyy.zzzz.gsm where xxxx is the agent number. However, if the queues use phone members (SIP/6000), the recorded filename is simply yyyy.zzzz.gsm. Is there any way of making the recorded file either agent-xxxx or even sip-xxxx where xxxx is the extension number. I had
2006 Jun 15
3
Auto-pickup cisco phones
Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get
2010 Jan 22
4
Snom vs Polycom
Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more "managerial" phones than the base phones which will be used for one line only. TIA Julian -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Oct 18
2
Agent recording and muxmon
I was wanting to use the new MuxMon application to record calls - it seems to be a "nicer" way of recording than the Monitor application. However, there is a slight issue with agents - we use the recordcalls option in agents.conf to record inbound agent calls - and I believe from looking at the source code that is uses the monitor application. Is there any way to get chan_agent to