Displaying 20 results from an estimated 10000 matches similar to: "How to return the status of a call to the calling server?"
2009 Aug 29
0
asterisk-users Digest, Vol 61, Issue 84
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'
2009 Aug 31
0
asterisk-users Digest, Vol 61, Issue 85
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
2010 Apr 20
1
Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]
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
>>>
2011 Sep 02
0
No subject
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> </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
2014 Mar 21
3
Need more meetme users -- hitting some limit
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
2010 Jan 03
0
asterisk-users Digest, Vol 66, Issue 4
"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
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http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
2005 Aug 23
2
SIP powercycle not hanging up
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"
2007 Apr 09
1
Re: asterisk-users Digest, Vol 33, Issue 35
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
>
2007 May 03
2
Balancing interrupts.
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
2015 Jun 26
0
Asterisk 13 logging to two places
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
2009 Jan 16
0
No subject
"In computer software standards and documentation, the term deprecation =
is=20
applied to software features that are superseded and should be avoided.=20
Although deprecated features remain in the current version, their use =
may=20
raise warning messages recommending alternate practices, and deprecation =
may indicate that the feature will be removed in the future. Features =
are=20
2015 May 17
0
Asterisk "virtual hosting"
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
2015 Apr 27
2
adding area code
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
2004 Sep 24
0
Re: Setting [rx/tx]gain for spandsp/fax
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,
2004 Dec 13
0
Transfer and keep variables
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
2005 Mar 07
0
iax2 setvars help needed
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
2005 Aug 03
0
chanspy not working with Agents
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:
2005 Aug 24
0
ANI2 AKA Info Digits not supported?
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
2005 Aug 27
0
how can I reduce delays in meetme with zap channels
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
2005 Aug 30
0
How to mute DTMF in meetme?
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