Displaying 20 results from an estimated 1000 matches similar to: "Is SIPPEER curcalls working for you ?"
2008 Nov 04
0
Is SIPPEER curcalls working for you ? [SOLVED]
2008/11/4 Igor Zamocky <asterix at ponozky.sk>
>
> Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers
> ?
>
I didn't.
Now I did and it's working the way I wanted.
Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and
SIPPEER but limitonpeers is much more concise.
Thanks a lot.
>
>
> Hi,
>
>
> In this
2013 Dec 02
2
creating opus encoder and decoder executable on Windows
Hi all:
How do I create opus encoder and decoder executable on Windows using MS Visual Studio 2010?
What I'm doing is the following:
1. I downloaded the codebase on my local drive using: $ git clone git://git.opus-codec.org/opus.git
2. Then I go to D:\opus\win32\VS2010 and open the opus.sln using MS Visual Studio 2010.
3. Then I build the solution opus. I get 3 executable at
2006 Mar 22
2
beronet & bristuff
Hi.
I'm trying to get a Beronet QuadBRI card work with bristuff drivers.
Though qozap module loads right, all card spans are in deactivated
status. I'm quite sure my configuration is correct and using a single
BRI card instead of the quadBRI the status is active and I can place and
receive calls.
On Beronet installation manual I read that Beronet and Junghanns cards
are identical in
2007 Feb 24
3
Mongrel Cluster 1.0.1.1 Prerelease: Healing power of the pack!
Hello all...
I know its been a while since the last one, but a new prerelease is
available with a gratuitous increase in version numbers!
gem install mongrel_cluster --source http://mongrel.rubyforge.org/releases/
This one has some new features in it that need to be tested on all unix
platforms. Please run --clean and cluster::status on every *nix you can
find. I run linux so the rest are up
2007 Nov 24
19
R/W lock portability issue
Hi,
I am having a problem porting Solaris R/W lock functionality to Linux.
The problem is that RW_LOCK_HELD() doesn''t have an equivalent function
in pthreads.
This was not a problem before because RW_LOCK_HELD() was only used in
ASSERT statements. In this case, I was able to make it work by making it
behave a bit differently (specifically, it would return true if *any*
thread was
2010 Jul 09
8
Repository of sample FXRuby programs?
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html><head><title>Repository of sample FXRuby programs?</title>
<META http-equiv=Content-Type content="text/html; charset=windows-1252">
<meta http-equiv="Content-Style-Type" content="text/css">
<style type="text/css"><!--
body {
2015 Mar 19
2
Brand new EATON 3S700DIN (mfr.date 09/28/14) doesn't wait for LB flag
Linux-2.4.28
libusb-0.1.8
nut-2.6.5 [+most recent drivers from GIT] - from tarball
EATON 3S700DIN [mfr.date 09/28/14]
----------------------------------
ISSUES:
I) Too many kernel logs:
---
In /var/log/kernel.log
usbdevfs: usb_submit_urb returned -28
In /var/log/daemon.log:
usbhid-ups[744]: libusb_get_interrupt: No error
usbhid-ups[744]: libusb_get_interrupt: error submitting URB: No space left
2008 Jun 11
3
Buffer in IceCast
Hi Tom,
You might be right, but I don't get why my existing downloads (I guess they
are on different routes) are not affected. If my ADLS provider is doing some
IP adjustments (btw my IP is static) I expect everything to disconnect, not
only my icecast connection. So I'll monitor my USRobotics logs today and
tomorrow so see if there is something strange.
Regards,
On Wed, Jun 11, 2008 at
2004 Oct 10
1
Fw: Souscription - Newsletter Ogg Vorbis en français !
----- Original Message -----
From: Claude de Limelette de Belgique
To: tarkin-dev-request@lists.xiph.org
Cc: vorbis-dev-request@lists.xiph.org
Sent: Thursday, October 07, 2004 9:15 AM
Subject: Fw: Souscription - Newsletter Ogg Vorbis en fran?ais !
----- Original Message -----
From: Claude de Limelette de Belgique
To: tremor-request@xiph.org
Sent: Thursday, October 07, 2004 9:13 AM
2008 Mar 11
2
AGI - calling functions, CHANNEL STATUS broken?
Greetings,
I am writing an AGI script that needs to check on the idle/busy status
of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and
Snoms thrown in for fun).
Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI
scripts? Based on my Googling, I would guess in the negative. I have
tried various permutations of Set() and Eval() without success.
I have also
2010 Jan 20
1
Using SIPPEER status with CUT function? SOLVED
On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote:
> Hi All,
>
> I'm using Asterisk 1.4 branch and checking the status of some SIP
> Peers with the functions ${SIPPEER(101:status)} and the result is "OK
> (48 ms)". ?Seems to work fine.
>
> Now I would like to use the function CUT to set a variable with the
> 'OK'
2010 Jan 20
1
Using SIPPEER status with CUT function?
Hi All,
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is "OK
(48 ms)". Seems to work fine.
Now I would like to use the function CUT to set a variable with the
'OK' portion of the status "OK (48 ms)" and then do some follow on
stuff if the status is OK.
I'm running into syntax
2009 Mar 25
1
SIPPEER equivalent for users.conf ?
Hi,
In sip.conf, it's possible to add a line such as
setvar=MYFIELD=foo
and access this value from diaplan with SIPPEER function.
1. Which function is available to access values in users.conf such as
vmsecret ?
2. Is it possible to extend users.conf with custom keys/values ?
Regards.
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2013 Dec 02
0
creating opus encoder and decoder executable on Windows
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html><head><title></title>
<META http-equiv=Content-Type content="text/html; charset=windows-1251">
<meta http-equiv="Content-Style-Type" content="text/css">
<style type="text/css"><!--
body {
margin: 5px 5px 5px 5px;
2008 Jun 11
0
Buffer in IceCast
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<html><head><title>Re: [Icecast] Buffer in IceCast</title>
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<style type="text/css"><!--
body {
margin:
2011 Dec 12
0
How to count ongoing calls from the dialplan
Hi,
When I need to route calls depending on the number of (incoming and
outgoing) calls a SIP device is currently handling, I mostly use
function SIPPEER and its curcalls option.
I can read and there references to function GROUP for the same usage,
but I intuitively thought that though this method also applies to
non-SIP devices and a large range of asterisk versions, it would
require more work
2009 Mar 16
2
Busy on SIP
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a call and then I try to call that phone, I find it
busy.
When SIP phone make a call and I try to call that phone, I find it avaible
and it rings but I want to find it busy.
I configure sip.conf like following:
[10]
type=friend
qualify=yes
2010 Jul 09
0
Installing FXRuby; error and solution
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html><head><title>Installing FXRuby; error and solution</title>
<META http-equiv=Content-Type content="text/html; charset=windows-1252">
<meta http-equiv="Content-Style-Type" content="text/css">
<style type="text/css"><!--
body {
2013 Oct 01
0
Feature request: SIPPEER or IAXPEER equivalent for DAHDI
Hello,
With setvar statements in chan_dahdi.conf, we have a convenient way to
store DAHDI channels specific values.
Unfortunately, we don't have a function to access this data from the
dialplan as easily as SIPPEER ou IAXPEER would for SIP or IAX trunks.
Using AST_CONFIG, you can access DAHDI setvar value but:
1. only one setvar value (see bellow)
2. AST_CONFIG reads values from current
2007 Jan 19
1
Red: Sip Phone CID
Here is what I have in my extensions.conf file now. Trustrcid and
sendrcid are turned to "yes" in the conf file.
I'm not fully sure how the SIPCalledRPID works though. The example I
found seems to try and provide the stuff automatically (id and name),
but does the SIPPEER stuff even exist?
I think this is probably the right track though. Any insight would be
much appreciated.