similar to: Play a number of files to a caller

I realize this is getting a bit outside myth...but hopefully someone can offer some ideas... I'm using the latest NVIDIA drivers on Fedora 12, with Nvidia 8600GT. Although the dual DVI outputs work great, the driver just won't detect anything connected to the component video connector. Is anyone out there successfully using the component video out on their Nvidia card with a recent

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

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 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.

why does this not work? i simply want to hear the recorded message exten => s,1,Answer() ;exten => s,n,Record(zipcodegutter1.gsm) ;zcg1 exten => s,n,Playback(zipcodegutter1) exten => s,n,Dial(SIP/c000001s/12222222259,120,A,(demo-thanks))

Using centos 5.2, I want to use a kickstart file to select packages in order to have an unattended install onto a bare-metal server. Is there any way of finding out which packages I need to compile, build and run asterisk ? I generally want to build all modules in asterisk and the wct4xxp zaptel module I want to be able to not select any groups in the kickstart file, but only select

Dear All, I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Thanks in advanced. Regards, Mehdi

Hello, I'm wondering if I can take benefits of long prompts to compute in the background the next step to be performed by Asterisk. Do you know what will be the behavior of asterisk if I send a STREAM FILE command immediately followed by another command ? Will asterisk stack commands or will it stop the first one to execute the second one ? If stops, is there any AGI library that handles

I cannot reach voip-info - is it just me or is the site not available ? Julian

I experience random crash of machine (full hang, requiring a hard reset) after trying to test run Asterisk 11. The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installed Anyone experience similar situation? -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

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 sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

I've opened a few bugs on ExternalIVR and added patches. The biggest issue is: https://issues.asterisk.org/view.php?id=16174 [patch] ExternalIVR does not handle arguments in a consistant manner Basically, this optimizes and fixes several different ways of calling ExternalIVR. If there is anyone who is using ExternalIVR today and/or is willing to test these patches I would appreciate the

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060

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

On Wed, Jun 6, 2018 at 6:18 AM, Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote: > > > Hi, > > > > I have tested ControlPlayback and grabbed files via an apache server with > > no issue. > > ControlPlayback is an Asterisk dialplan function. > > How have you integrated this

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

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