Hi all,
I can't find the answer to this via google - is there some way to
permanently enable sip set debug on and agi set debug on in Asterisk? I
want this to be automatically enabled even after restarts.
Thanks for any advice.
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free:
On Tuesday 08 November 2011, Yaroslav Panych wrote:
Greetings
I have found next paragraph in Licence file(source root)
Digium, Inc. (formerly Linux Support Services) holds copyright
and/or sufficient licenses to all components of the Asterisk
package, and therefore can grant, at its sole
On Nov 8, 2011, at 9:55 AM, isr...@gmail.com wrote:
There is a bug which blocks call progress message 8 which was fixed
but I don't remember in which version
Try upgrading to latest 1.6 version
Before we opened for the day today I updated to 1.6.2.20 and that
seems to have solved the
Is anybody using pci-passthrough?
2011/11/9 Nick Khamis sym...@gmail.com
Hans,
Thank you so much for your response. We will be moving everything to VM
soon.
Cheers,
Nick.
On Tue, Nov 8, 2011 at 6:11 PM, Hans Witvliet aster...@a-domani.nl
wrote:
On Mon, 2011-11-07 at 11:45 -0500,
On 11/09/2011 04:22 AM, David Cunningham wrote:
Hi all,
I can't find the answer to this via google - is there some way to
permanently enable sip set debug on and agi set debug on in
Asterisk? I want this to be automatically enabled even after restarts.
In recent versions of Asterisk, you can
Hahah... I was waiting on the sideline for this question.
Nick.
On Wed, Nov 9, 2011 at 8:10 AM, Anton Kvashenkin
anton.juga...@gmail.com wrote:
Is anybody using pci-passthrough?
2011/11/9 Nick Khamis sym...@gmail.com
Hans,
Thank you so much for your response. We will be moving everything
On 11/08/2011 07:54 PM, Raj Mathur (राज माथुर) wrote:
On Wednesday 09 Nov 2011, Kevin P. Fleming wrote:
[snip]
* The GPLv2 places no restrictions on what you can 'write', it only
places restrictions on your distribution of things that you write
that could be considered 'derivative works' of a
On 11/09/2011 04:37 AM, A J Stiles wrote:
On Tuesday 08 November 2011, Yaroslav Panych wrote:
Greetings
I have found next paragraph in Licence file(source root)
Digium, Inc. (formerly Linux Support Services) holds copyright
and/or sufficient licenses to all components of the Asterisk
package,
I shall contact when(and if) decision will be made. But such decision
cannot be made basing only on this paragraph, because it does not
describes anything. There are no description of licensing procedure,
nor pricing, nor liability, rights or freedoms(at least in general
approximation) of sides.
On 11/09/2011 07:59 AM, Yaroslav Panych wrote:
I shall contact when(and if) decision will be made. But such decision
cannot be made basing only on this paragraph, because it does not
describes anything. There are no description of licensing procedure,
nor pricing, nor liability, rights or
On Wednesday 09 Nov 2011, Yaroslav Panych wrote:
I shall contact when(and if) decision will be made. But such decision
cannot be made basing only on this paragraph, because it does not
describes anything. There are no description of licensing procedure,
nor pricing, nor liability, rights or
On Wednesday 09 November 2011, Kevin P. Fleming wrote:
On 11/09/2011 04:37 AM, I wrote:
What you *can't* do is distribute your modules *as pre-compiled binaries*
under any licence beside the GPL -- if they are distributed under any
other licence, they *must* be compiled on-site by the end
But so long as you were careful not to copy any of the code you are
going to link against into your Source Code (and why would you, if
you were linking against it?), it only *becomes* a derivative work
*after* it has been compiled.
That's not necessarily true because if you have a work that
As promised, here is a follow up on my quest to get CallerID
correctly
presented when forwarding calls to cellphones.
Here is a reminder of the issue at hand:
Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
which forwards to Cory (GSM handset)
What I
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins,
so in the example below, for example, how can I get the current user count
in ConfBridge 3000?
[conferences]
;authenticated conference (ext
2011/11/9 Richard Mudgett rmudg...@digium.com
As promised, here is a follow up on my quest to get CallerID
correctly
presented when forwarding calls to cellphones.
Here is a reminder of the issue at hand:
Alice (GSM handset) calls Bob (ISDN-connected Asterisk
2. As I feel specically new to this RDNIS concept, how should I set
CALLERID(RDNIS), before or after Answer() statement ?
It does not matter in this case. Asterisk v1.6.1 will keep both legs
of the call anyway.
If you ultimately want to get the call entirely off of your Asterisk
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference
identifiers
(0 or 1 for 'locked')
Unfortunately, that function doesn't seem to be in 1.6.20, which
Asterisk version are you using?
*CLI core show function CONFBRIDGE_INFO
No function by that name registered.
Command 'core show function CONFBRIDGE_INFO' failed.
On Wed, Nov 9, 2011 at 12:24 PM, Danny Nicholas da...@debsinc.com
10.0.beta2. Have you tried confbridge(xxx,c)? This joins and announces
count, but I don't know if it returns a variable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday,
confbridge(xxx,c) is a blocking call, so you can't get status back
until that command completes. Time to upgrade to 10.0.beta2 I
guess...
On Wed, Nov 9, 2011 at 12:47 PM, Danny Nicholas da...@debsinc.com wrote:
10.0.beta2. Have you tried confbridge(xxx,c)? This joins and announces
count,
What about a local call to confbridge(xxx,c)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 12:57 PM
To: Asterisk Users Mailing List - Non-Commercial
Kevin,
Thank you very much!
On 10 November 2011 00:15, Kevin P. Fleming kpflem...@digium.com wrote:
On 11/09/2011 04:22 AM, David Cunningham wrote:
Hi all,
I can't find the answer to this via google - is there some way to
permanently enable sip set debug on and agi set debug on in
If you have an ancient version of Asterisk you want to stick with, you can
do this with asterisk -rx sip set debug on and asterisk -rx agi set debug
on in your safe_asterisk script.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote:
Is anybody using pci-passthrough?
Yes, though quite a while ago.
About three years ago, i used pci-passthrough to give a dom-U access to
a localy mounted smartcard.
But i have a vague feeling that you are up to something else...
I know
Smart card? I think we should be leaning more towards the network devices?
Cheers,
Nick.
On Wed, Nov 9, 2011 at 5:23 PM, Hans Witvliet aster...@a-domani.nl wrote:
On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote:
Is anybody using pci-passthrough?
Yes, though quite a while ago.
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http://au6vpf8so.blog.com/1d/
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--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi All,
I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in
routing calls to upstream carrier via SIP trunks out. I spent a lot of time
in the lab testing 1.8 which included heavily testing DTMF with no issues
that came up. It all just seemed to work fine. But then
I had similar problems with 1.8.6 and polycom phones intermittently having
DTMF issues. I updated to 1.8.7 and things cleared up. I went through the
release notes at the time, but don't recall which commit made me decide to
give it a try.
Rgds,
Jared
On Wed, Nov 9, 2011 at 7:03 PM, JR Richardson
Hi folks,
What methods are available for testing IAX2 service availability? I
know about iax2 show peers and iax2 show registry, but I'd like
some alternatives.
Tcpdump shows a little more about what's going on, but a handy test
using nmap doesn't seem to work anymore (see
I would recommend you Monast (monitor asterisk), it's stable and gives
alot of information.
http://sourceforge.net/projects/monast/
WBR
A.Rymkus
04.11.2011 13:34, Anthony Laudini ?:
Hi Jean,
I suggest Queuemetrics. There are many out there but this one is good
for monitoring and
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