similar to: Asterisk question

not able to run my php from AGIi am using asterisk 1.8.13 (debian)i am able to make call file using php command line..but when executing php from AGI, it is not working..kindly see the attachment if bellow text is not readable...___________________________________________________ File: /etc/asterisk/extensions.conf[call]exten => call,1,Answerexten => call,n,Playback(hello-world)exten =>

I am interested in testing two similar nls models to determine if the lines are statistically different when fitted with two different data sets; one corn, another soybean. I know I can do this in linear models by testing for interactions. See Introductory Statistics with R by Dallgaard p212-218 for an example. I have two different data sets I am comparing to lai. ci.re should have very

Anybody was able to set it up?? I can't make it work, any idea?? Ast 1.6.0.1 Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081125/c7f6be25/attachment.htm

I am attempting to forward http requests to my external interface, from internal machines to a machine that is located on the internal interface, via the firewall rules. Externally, I am able to forward the port to the webserver located behind the firewall, and I want to use the same hostname/ip for clients if they are on both sides of the firewall. Note, that I only want to do just the one port,

Someone on this list uses the address @sedwards.com I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either. Part of my mail servers reject these emails because they cannot be replied to, or are likely to be spam. Every so often I get a mail from the list management to say that I've been unsubscribed

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

libswfdec/swfdec_as_interpret.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletion(-) New commits: diff-tree 38fbc1389267e593b44041018cbb1750bdcce0fb (from aaca94203d8a0ccb8feb32c0d57df3401fca0350) Author: Benjamin Otte <otte at gnome.org> Date: Sun Jun 17 14:19:45 2007 +0200 actually convert the values to a string when comparing strings diff --git

I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance,

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

> On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards <asterisk.org at sedwards.com> wrote: > Now that the g729 patents have expired, how do we use g729 in > Asterisk? > > Will Digium be releasing a g729 codec for 'free' use or do we > download the 'free' codec off the Internet now that we can use it > without moral or legal

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

Now that the g729 patents have expired, how do we use g729 in Asterisk? Will Digium be releasing a g729 codec for 'free' use or do we download the 'free' codec off the Internet now that we can use it without moral or legal restrictions? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

Hi All, I have installed astriesk 6 and am able to make calls using sip x-lite. Its working as I expected. Now I want to make call from sipx-lite to PSTN using asterisk. can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.). 2) what is that I need to do after buying the card to make it talk to the real world PSTN network?

On Mon, 18 Jan 2016, Ethy H. Brito wrote: >> how to flush user input before READ()? How about a read() to a dummy variable with a 1 second timeout to consume the octothorpe and password? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

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

I finally found a reason TO run Asterisk as root. By default, ext[23] file systems "reserve" 5% of the filesystem for root. Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call