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On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

* The exit behavior of the AGI applications has changed. Previously, when a connection to an AGI server failed, the application would cause the channel to immediately stop dialplan execution and hangup. Now, the only time that the AGI applications will cause the channel to stop dialplan execution is when the channel itself requests hangup. The AGI applications now set an AGISTATUS

Wrong list. asterisk-dev is for changing the C source code of Asterisk. I don't think AGI's "count" or are considered for inclusion into the subversion repository as stated by one of your conditions for payment. On Wed, 29 Apr 2009, Alistair Cunningham wrote: > I'd like to offer a bounty for a feature for Asterisk where an AGI > program can park and retrieve calls

I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? -- Thanks in advance,

Topic 6: RE:unable to execute command hi there i tried to execute the command as suggest like exten => 1987,1,Playback(posix-restarting) exten => 1987,2,wait(1) exten => 1987,3,System(/usr/bin/python /home/docas/Desktop/mess1.py) exten=> 1987,4,Hangup it still doesn't work,now it does it give unable to execute command but it doesn't reach the system command it just

core show function SIP<TAB> I use: set(PEERIP=${SIPCHANINFO(peerip)}) in one of my dialplans. For AGI, whatever function in your library that executes 'GET FULL VARIABLE' should do the trick. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline

is typing his number, though there is a 15 seconds timeout, and even if = I type the number very fast it still may happen to me.<o:p></o:p></p><p = class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>It has = been my casual observation that the speed at which I enter digits on my = phone is unrelated to the speed at which my

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

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"

Wrong list. asterisk-dev is for changing the C source code of Asterisk. That's part of why you didn't get a response yesterday. On Wed, 29 Apr 2009, Juan Miguel Quiros Arrieta wrote: > I have to develop an application using the VeriFone vx510 device and I > read this device needed or could use a PPPoE connection in order to > validate and send all information collected from

Trying to redirect to -user... On Tue, 7 Jul 2009, Bryant Zimmerman wrote: > Hey guys I posted this earlier and did not get any responses. You posted what appear[s|ed] to be a user question to the dev list. I did reply (on June 3), but I may have mis-understood. > I am working on some AGI development that requires control of audio file > playback. The control stream file is working

Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. I'm googling now and I recognize that Friday afternoons are the worst time to ask questions, but I'm getting desperate because this is keeping me from rolling a system out to production. (Yup, I know. Who rolls out a system on a Friday

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

On Sat, 8 Aug 2009, Max Alex wrote: > Actually the scripts which are set to run to the hangup of channels, > which is originated for sending fax. We are trying to get the answer > time, duration of fax on hangup of that channels, but the script becomes > stuck and we need to restart the asterisk and also we are not getting > any output of script as it is stuck. Let's start

I'm trying to pass a variable between servers using "setvar" in iax.conf. I have a box (ts2) with a t100p in it. It answers the call and dials another box (ast0) via IAX. I want to pass a variable along with the call from ts2 to ast0. I'm running CVS-HEAD-03/07/05 on ts2 and ast0. ts2's iax.conf: [general] disallow = all allow

On Fri, 26 Jun 2015, Dale Noll wrote: > I added a filter to the /etc/rsyslog.conf file > > :syslogtag, contains, "asterisk" stop > > Syslog is still receiving the messages, but is discarding them. Nice to learn a new (to me) feature of rsyslog. What does 'logger show channels' show? -- Thanks in advance,

On Fri, 10 Jul 2009, Jared Mauch wrote: > I need the 'talking' information to better identify rogue people > on bridges. I'm a 1.2 Luddite so I don't have all these fancy new features :) A different solution to a similar problem. I had problems with abusive callers in my conferences. I whipped up some dialplan and AGI mojo to let an admin mute and unmute individual

On Mon, 27 Apr 2015, Bryant Zimmerman wrote: > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

I've googled for way too long, where are the 2 letter language values defined? I know: en = English es = Spanish fr = French but what about Croatian, Russian, Serbian, Vulcan, etc? Is there a list documented for Asterisk or is it "just use the 2 letter country code Internet TLD?" Thanks in advance, ------------------------------------------------------------------------ Steve