similar to: Asterisk 1.2 crashing on x64 when transferring a call

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,

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

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 8/27/10, asterisk-users-request at lists.digium.com <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

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'

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

Is there any way to transfer a call from host to host and keep the call's variables intact? -- specifically, UNIQUE_ID and user created variables like CARD_NUMBER, EXPIRATION_DATE, and CVV2? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline

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 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"

"asterisk-users-request at lists.digium.com" <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' to

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 >

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

I see the following on one of my new servers: -ts10::sedwards:~$ cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 2979045 2988620 87780075 87779501 IO-APIC-edge timer 1: 1 3 2 3 IO-APIC-edge i8042 8: 0 0 0 1 IO-APIC-edge rtc 9: 0 0 0

I turned on the messages that he had in the file again, all the logs were in /var/log/asterisk and it does not show anything for syslog. asterisk -rx 'logger show channels' Channel Type Status Configuration ------- ---- ------ ------------- /var/log/asterisk/full File Enabled - DEBUG NOTICE WARNING

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

On Sun, 17 May 2015, martin f krafft wrote: > also sprach Steve Edwards <asterisk.org at sedwards.com> [2015-05-16 23:22 > +0200]: >> I use a preprocessor >> (http://software.hixie.ch/utilities/unix/preprocessor/) to tailor >> dialplans and configuration files to each host based on the client (or >> project) and the hostname. On Sun, 17 May 2015, martin f

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

I was debugging a turnup with Global Crossing the other day and they presented me with a web page that displayed a 'ladder diagram' of a call including a ton of detail all neatly organized in tabs and links so you could drill down to any level of detail needed. The copyright notice says 'Copyright? 2008 Empirix.' Is there any free software available to analyze a pcap or

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,