similar to: Redirect two channels to each other?

I've been using asterisk for 3+ years now, I love it, but it doesnt love me back. :-) It was crashing frequently and seemingly randomly prior to this latest upgrade. Not sure what version it was running prior to upgrade (it was probably an old CVS HEAD from 2+ years go.) Anyway, currently running 1.4.21.2. == Problem == Problem is that its crashing for seemingly no reason at all, no

2008/9/29 Alex Balashov <abalashov at evaristesys.com> > Try this: > > exten => _XXXX,1,Set(THISTECH=${CUT(CHANNEL,/,1)}) > exten => _XXXX,n,NoOp(Technology is ${THISTECH}) > exten => _XXXX,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)}) > exten => _XXXX,n,NoOp(Channel is ${THISCHANNEL}) Hi, I don't have any spare zaptel enabled system I could try this on, but I

Dear all, I'm stuck with a problem using plyr to process a rather large junk of data. What I'm trying to do is applying a moving average to all the subparts of the dataframe (the example data can be found here https://dl.dropboxusercontent.com/u/2414056/testData.Rdata). require(plyr) load("testData.Rdata") applyfilter<-function(x){ return(filter(x,rep(1/5, times=5))) }

Hi all, I know when doing a Dial, when the called party hangs up, we have a few different ways to redirect the calling party to other parts of the dialplan. In this case, I have someone who would like to do the opposite... When the calling party hangs up after a Dial(), redirect the called party to another location. I'm not sure how else to describe what the user wants to do, but I'm

Hi folks ! I bought two sipura 841 phones. I used to have GN Netcom headset which I connect instead of the handset. The problem is that I don't have any sound coming out the headset and I can't speak neither ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to

Hello, I am a newbie in Asterisk IP PBX but I am very impressed by its functionalities. I have read that It can work over IP network and across the PSTN. I am not very sure how it works over the PSTN.. In case if people have not yet the Internet or SDL access, I would like just to know if it is possible to interconnect two IP LANs using the traditionnal analog Network and Asterisk PBX? In such I

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

> > Thanks Josiah Bryan, > > I do not have any dns server running on my asterisk server, we do have an > external DNS server working in the data center; the IP of this dns server is > 10.4.1.5... > > Following you will see my main configuration: > > /etc/resolv.conf: > > domain localdomain > search localdomain > nameserver 10.4.1.5 > nameserver 10.4.1.2

Hey all - Attached is quick perl script i whipped together this morning to dial from Kontact (well, KAddressBook). You'll need Net::Telnet (perl -MCPAN -e 'install Net::Telnet') for it to work. You'll also need to make sure you've got a valid user/secret in /etc/asterisk/manager.conf on your * server for this to work. Usage: /usr/bin/asteriskdial --from

Hi! I'm building an application which allows to dial via the Asterisk Manager Interface using the originate command. There should be an optional conferencing feature. The manager commands are basically: --------------------------------- action: login username: sdjklgdsjg secret: xxx events: on action: originate callerid: 3847438609 priority: 1 exten: 4068439865 async: 1 context: out

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconnected. Is this the expected behavior? Is there

Is there any way to end up somewhere in between the development environment and production environment as far as refreshing the source on every request. Our setup is that we work off of nfs shares which makes it extremely slow for a page load in development since it has to pull in the whole app on each request. I''m used to it by now and only get a few changes in an hour :). Just curious

Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way to redirect one of the channels and hold on to the other. Thanks, Neil Cherry

You described it well; I?m afraid that?s what it is. They want documents of an organization that doesn?t fit the description, and only then will they test it. That sounds great. I?ll reach out to Jeremy and see if he can help me get the IDs the USAF wants. It sounds like Joshua predates my time here. That?s a really good idea. Thanks very much! Doug From: Josiah Parry

I'm trying to figure out how to use virsh (or something else) to list all the IP addresses (or MAC addresses if needed) for each virtual NIC, preferably with it domain affiliation also listed. Is this possible? Thanks, JSR/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm wanting to take an existing call, and transfer both sides of it into a meetme room (yes I know the phones have a conference ability built-in but humor me). What seems to happen is I can transfer one half of it fine, but as soon as I do that the other half hangs up. Do I have to park it briefly? If so, what does the call ID become once it's parked, so that I can

Hey Doug, R is not a product that is provided by a company or any vendor that can be procured through a vendor e.g. something on a GSA schedule. Seems like you're caught in the bureaucracy hell hole. I used to help the USAF, and other DoD members use R when I was at RStudio (now Posit). I recommend you find someone in your organization who is doing Data Science. They'll likely have

Not sure if this is the right place to come to, but I don't have "RHN" support.....I'm hoping someone can help me out here. I have downloaded the RHEL .ios file and burned it to DVD/CD, my laptop is primed to Boot From CD/ROM Drive, I start the installation using the semi-graphical interface, and the first few options are a breeze, then it gets to identifying the CD drive