similar to: Transfer and keep variables

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'm wondering if tweaking [rx|tx]gain would improve my fax reception success rate. Running ztmonitor when receiving a fax shows 4 "octos" and an * on the RX side and nothing on the TX side. At the end of the page, there's a burst where RX goes to about 1/2 and TX goes to about 2/3 of the range displayed. Any opinions? Thanks in advance,

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

I'm trying to spy on an agent (Agent/54321). I can "dial(Agent/54321)" successfully. If I "chanspy(Agent/54321)" or "chanspy(Agent)" all I get is a series of beeps. Any clue where I should start looking? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice:

I'm not receiving ANI2 (info digits) on my SBC PRI's. SBC said they're sending them. I called Digium support and was told it is not supported. Is anybody receiving ANI2 on a PRI? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline pagesteve@sedwards.com

My boss is complaining that the delay between speaking and hearing in a meetme conference is noticeable and doesn't want to roll out our system until I can eliminate the delay. Personally, I don't think the delay is significant, but I don't sign his check. The system consist of 3 1u's, each with a single quad t1 card. Each card has 2 t1's running NFAS. The "t1

This is weird. If I have 2 members call into meetme using zap PRI channels on the box, they can here each other's keypresses. If I have 2 members call into a separate box using the same PRI's and then forward (dial(iax/...)) them to the previous box into the same meetme, they only hear a minor "squelch" for each other's keypresses. How can I completely mute a

A lot of people are going for the "VOIP only" approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com

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

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

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

I want to create a voip service, I do not know much about it, but the first thing I want to know if more than one client can make a call at the same time through internet to the PSTN, and what gateway should I use for this. 2014-12-08 13:07 GMT-08:00 Steve Edwards <asterisk.org at sedwards.com>: > On Mon, 8 Dec 2014, Leonel Florin wrote: > > Hay friends, I want to know how many