similar to: Looking for free DID with IAX

Steve, These are live streams of events so I can't simply rip the audio. As I mentioned at the end of my email putting in a sleep did help a bit however there are only so many streams Asterisk will grab nicely at once with out spiking the CPU. I also tinkered a bit with real time here is what I found: 1) If we have cachertclasses=no then Asterisk will only pull the stream if some one is

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

Thanks for the feedback ! I know the IP-address of my Asterisk-server. The WAN-interface of my Asterisk-box is set manually (ifcfg-eth1). I have port 4569 forwarded on my NAT/firewall. Strangely I have the same 'notice' when being attached directly to the internet (so no firewall in between). And set my WAN-interface to the IP I get from my ISP or even when obtained by DHCP. Doesn't

This may be of interest -- as a tool we can use to test our systems and as a weapon that may be used against us :) http://warvox.org/ A brief read-over looks like it uses iaxclient and ruby to war dial a range of numbers and record audio samples to be analyzed to identify if the call was answered by a modem, fax machine, human, etc. The calls are placed through a PSTN termination

I'm trying to setup a pair of FreePBX-4.211.64 boxes using Pacemaker, Corosync, and DRBD. All the examples I've found so far use Heartbeat, but Heartbeat is not in the repositories and doesn't want to compile from source. Does anyone have a working configuration they can share or a tutorial they can point me to? Also, what does drbdlinks bring to the party? Isn't just linking

Does anybody have contact info for Dan Toma, the author of Diax? I've tried danto at clicknet.ro and danto at rdslink.ro without success. Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

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'

Hi, Here is a general question, not applying to asterisk so much, but in the application of asterisk. I have purchased a few IAX DID's through VoicePulse and am interested in a service provider who has the ability to provide me with one number (reliable, as I wish to publish), and the capacity to redirect those calls to my IAX DID's (is this even possible)? Also, with IAX

I have a client that wants to put a phone on their web page for customers to call them via their Asterisk server. ) A keypad is needed to enter credit card details. ) "Speed dial" buttons like "Tech Support," "Sales," etc. are a requirement. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a

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

"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

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"

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

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 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 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

On Tue, Apr 20, 2010 at 10:49 AM, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Tue, 20 Apr 2010, Tilghman Lesher wrote: > >> On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote: >>> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI> >>> prompt, and found references on using the command "soft hangup >>>

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