Le 08/02/2012 23:28, Kevin P. Fleming a écrit :
On 02/08/2012 04:02 PM, Danny Nicholas wrote:
Not a complaint, per se, just a question. Why are the LTS versions
odd
(11, 13, 15, etc) and the non-LTS (10, 12, etc) even? As I read the
chart,
Digium/Asterisk is committing to a new LTS version
Le 09/02/2012 09:49, Administrator TOOTAI a écrit :
Le 08/02/2012 23:28, Kevin P. Fleming a écrit :
On 02/08/2012 04:02 PM, Danny Nicholas wrote:
Not a complaint, per se, just a question. Why are the LTS versions
odd
(11, 13, 15, etc) and the non-LTS (10, 12, etc) even? As I read
the
2012/2/8, Carlos Alvarez car...@televolve.com:
If the customer is so cheap that they won't properly build out the network,
why would they have gigabit switches to the desktop which have a limited
set of applications that actually benefit from it?
Then there's PoE, which is expensive to start
Hi All,
This may be an off topic but I'm not sure who else would know the answer. I'm
playing around with Asterisk and Polycom phones. I see Polycom supports quite
a few codec. The usual ones and these:
G722
Siren14.24kbps Siren22.32kbps
Siren14.32kbps
2012/2/8, Kevin P. Fleming kpflem...@digium.com:
On 02/08/2012 12:40 PM, Olivier wrote:
2012/2/8, Kevin P. Flemingkpflem...@digium.com:
On 02/08/2012 10:06 AM, Carlos Alvarez wrote:
On Wed, Feb 8, 2012 at 2:35 AM, Olivieroza_4...@yahoo.fr
mailto:oza_4...@yahoo.fr wrote:
I always
Only the higher end Polycoms support Siren7 and Siren14. I believe only the
VVX and SoundStation IP phone support those codes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn
Sent:
On Wed, 8 Feb 2012 18:17:46 -0800, Chad Wallace
cwall...@lodgingcompany.com wrote:
Maybe the release announcements are what you're looking for. e.g.,
for 1.8:
http://www.asterisk.org/node/51444
And you can probably find the same for 1.4, 1.6.x, and 10 without too
much trouble.
Thanks. It's
On Wed, 08 Feb 2012 20:23:54 -0600 (CST), Richard Mudgett
rmudg...@digium.com wrote:
The CHANGES file is not just a dump. It is a manually created file that
documents each feature addition. There is a ChangeLog file that is a dump
of every single commit made to the source file.
Sorry about
On 9 Feb 2012, at 11:08, Gilles wrote:
Does someone of a good site/blog that keeps track of new releases of
Asterisk, and explains what the major changes/features when they do
occur?
Why not just use the latest version?..
S
--
On Thu, 9 Feb 2012 11:13:38 +, Steven Howes
steve-li...@geekinter.net wrote:
Why not just use the latest version?..
Because converting Asterisk to run on that non-x86 platform is quite
some work, so I need to know what I'm missing by staying with a 1.4.x
release.
--
Hi everybody
We're facing a strange problem with Asterisk (1.8.2.3) executing an AGI. The
script (python) is updated and after a invocation by the dialplan, the new code
is executed but the ${AGISTATUS} variable shows the wrong value. The right
value only appears if a 'core reload' is executed
Am 07.02.12 12:38, schrieb virendra bhati:
Hi List,
Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What
technology FreeSwitch is used and asterisk don't. I don't know it's the
right or wrong but this question come to my mind...
I had done some load tests with asterisk
Wow,
I bet even asterisk developers wouldn't believe so. What have they done !.
No, actually can you tell if server was processing media along with the
calls as well !?
I once tested without media and really I had some 1000+ CCs on asterisk
server on a regular dev machine with choppy audio on an
Hello Shaun,
I take this approach :
- zaptel 1.2.XX , 1.4.XX I can see interrupts working , I made some
patches to compile in 2.6.32 kernel tree.
- dahdi 2.0.0 to dahdi-2.2.0, I can see interrupts working, I made some
patches to compile in 2.6.32 kernel tree
- This itop from 2.2.0 tree
From: Stefan Schmidt s...@sil.at
Sent: Thursday, February 09, 2012 6:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch
Am 07.02.12 12:38, schrieb virendra bhati:
Hi List,
Why FreeSwitch can handle more
just done the test again.
13500 concurrent calls at 1750 cps with open rtp ports but without much
media transportet, only signaling. see attached screenshot.
1 concurrent calls with media playing musiconhold but i only have a
100mbit connection on this server so i cant do more here.
the
From: Stefan Schmidt s...@sil.at
Sent: Thursday, February 09, 2012 8:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch
just done the test again.
13500 concurrent calls at 1750 cps with open rtp ports but
Am 09.02.12 14:19, schrieb Bryant Zimmerman:
Stefan
This is on target with my configuration I am working on. What kind of
dialplan were you using when running the tests.
Were you doing database lookups or just answering the calls and playing
hold music. Any example would be
Hi,
Sometimes some of my dahdi channels become stuck, It is very
strange, here is the output of the core show channels command:
pabx*CLI core show channels
Channel Location State
Application(Data)
Local/104@ramais-cc0 104@ramais:1 Up
If you do a core show channel conicse and use the id from the LOCAL side
of the call that is stuck that should shut down the channel without
clearing all your DAHDI channels that are up. Does it only happen when you
are calling throught a LOCAL context?
Thanks
Bryant
hi,
We have a phone number from third party provider which is used for inbound
calls. How could I monitor if this *phone number* is reachable?
the initial idea doesn't sound elegant:
- on my SIP server I set couple seconds of ringing before Answer().
- the monitoring server calls to that phone
I would probably set the calling server with a specific callerID,
match on that caller ID on the recipient server and then log to a
database or trigger a script to import my data. By doing that I can
confirm the carrier side of things are up and that my server is actually
processing calls. With
Hi,
I'm also interested in rpm packages including chan_gtalk and res_jabber
because I do not want to have a build environment on my productive server.
Does anybody knows the reason why this is not available via rpm?
best regards
Christoph
Am 28.11.2011 05:30, schrieb Vladimir Mikhelson:
I
We designed our solution the following way.
We have several land line numbers hooked to an asterisk testing server.
The testing server places one call every X seconds per line to a number we
want to test . We cycle through each number in our testing pool. Each
number on average is tested once
On 02/09/2012 04:08 AM, David Klaverstyn wrote:
I can get Asterisk to work with G722 and the sound is superior compared
to uLAW. I tried to get it working with Siren7 and Siren14 but I cannot.
It always says incompatible codec and what is this Siren14 and Siren22
with Polycom. Is this different
Hi everyone,
I have tons of CDR from an Asterisk with a PRI connection. I want to know
som extra details about the calls like the maximum number of calls in peak
hours, etc...so I am looking for a php or other type of script that would
show this to me in a GUI graphica format. Ideally, it would
On 09-02-12 14:52, Stefan Schmidt wrote:
Am 09.02.12 14:19, schrieb Bryant Zimmerman:
Stefan
This is on target with my configuration I am working on. What kind of
dialplan were you using when running the tests.
Were you doing database lookups or just answering the calls and playing
hold
2012/2/9 Antonio Modesto mode...@isimples.com.br
**
Hi,
Sometimes some of my dahdi channels become stuck, It is very strange,
here is the output of the core show channels command:
pabx*CLI core show channels
Channel Location State
Application(Data)
From: Patrick Lists asterisk-l...@puzzled.xs4all.nl
Sent: Thursday, February 09, 2012 10:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch
On 09-02-12 14:52, Stefan Schmidt wrote:
Am 09.02.12 14:19, schrieb
Am 09.02.12 16:45, schrieb Patrick Lists:
Iirc a long time ago there was a discussion about load testing by
playing MoH was not a realistic test. Something about all MoH music
getting streamed synchronized so basically Asterisk only has to stream
one file and sorta multiplex that single output
Hello Shaun,
1) dahdi-linux-complete-2.2.0.2+2.2.0 , is generatiing 1K int/s after
patches..
2) Looking in 2.2.1 code, i see to wct4xxp.c the alarmdebounce set in
2500 and in 2.2.0 the alarmdebounce seted to 0,
I load the module with alarmdebouce=0 paramter = the interrupts are up
to 1000
Hi,
I'll try to havebeen help for asterisk AMD module.
Sorry for my bad english but I'll try to speak the better I'll able to do.
So, here's my project:
I did IVR.
If you're pressing 1 key, asterisk's calling a mobile phone line.
During the ringing, asterisk is lunching musiconhold for
A very interesting solution. Is there any code you'd share for this?
We don't have inbound issues all that often (as far as we know), so I'm
curious whether you had a lot of reliability issues before this, or
possibly we have more problems than we believe.
On Thu, Feb 9, 2012 at 7:59 AM,
Brilliant! `log the call and busy out` is the thing I was missing.
thank you so much
On Thu, Feb 9, 2012 at 4:59 PM, Bryant Zimmerman brya...@zktech.com wrote:
We designed our solution the following way.
We have several land line numbers hooked to an asterisk testing server.
The testing
Our Asterisk system (1.8.8.1-1digium1~squeeze) has been very
stable and generally doing a good job -- except that one day,
voicemail recordings started being garbled.
But wouldn't that mean that every customer line is busy every 30 minutes
for a few milliseconds for real callers? Unless there is more than 1
concurrent call enabled on the customers line.
:)
Am 09.02.2012 15:59, schrieb Bryant Zimmerman:
We designed our solution the following way.
We have
Reply to self, missed the line count part. Nevermind then :)
Am 09.02.2012 18:10, schrieb Markus:
But wouldn't that mean that every customer line is busy every 30 minutes
for a few milliseconds for real callers? Unless there is more than 1
concurrent call enabled on the customers line.
:)
Hi Dan,
my wild speculation: It's some kind of timing/synchronisation problem.
Do you use jitter buffer an/or echo cancelation?
Best regards,
Ruben
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Dan
From: Carlos Alvarez car...@televolve.com
Sent: Thursday, February 09, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] checking if a phone number is UP
A very
Thanks for the detailed followup. Inbound reliability improvement is on
our 2012 goals.
When you place your outbound test call, are you mindful of the carrier it
goes on, do you vary them, etc? In other words, do you do anything to be
sure that while carrier X can place a call to carrier Y, you
Hello,
I know about the german phone system that the sense of an anonymous call
is, that the called party has no way to get the caller's number in any way.
The last switch honours the anonymous bit and removes the phone
numbers before sending the call to the called party.
In EURO-ISDN you have a
Markus
No we do checks ahead of line count checks in the dialplan code.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Markus unive...@truemetal.org
Sent: Thursday, February 09, 2012 12:08 PM
To: asterisk-users@lists.digium.com
Hi,
on a similar setup I set in sip.conf:
prematuremedia=no
progressinband=never
in the peers configuration.
With this config you tell Asterisk not to handle inband information at
all. But: Maybe you won't get any inband error messages also.
Greetings from Wuppertal
Max Grobecker
Am
We have used in production Asterisk 1.4 to do 3,000 concurrent calls at
about 80 CPS without media going through the system. This is on a vmware
ESXi server. The server is a Dell R610 with 2 X5670 (6 cores each at 2.93
GHz so 12 physical, 24 logical cores). Each vm gets 2 cores and 2 GB of
RAM. We
On Thu, Feb 09, 2012 at 02:11:12PM -0200, Marcio Gomes wrote:
Hello Shaun,
1) dahdi-linux-complete-2.2.0.2+2.2.0 , is generatiing 1K int/s
after patches..
2) Looking in 2.2.1 code, i see to wct4xxp.c the alarmdebounce set
in 2500 and in 2.2.0 the alarmdebounce seted to 0,
I load the
I am having a strange problem with an external SIP phone. It can
register and receive calls but it cannot initiate any calls. A
softphone on the same network works without problems.
As far as I can notice the difference is that the hard phone is not
sending the proper contact
If the MOH thing is really true, a more realistic test would be to run
playback(demo-instruct). Since I know that I will eventually cross this
bridge in real life/real time, I devised this test on my Asterisk 10.0 box
Dialplan (in default context)
exten = 3366,1,answer()
exten =
Hello,
I've installed the free (one user) Fax for Asterisk (FFA) license.
Outgoing faxes, using T.38, to the PSTN work quite well. However,
incoming faxes, do not seem to detect tones and certainly do not switch
to T.38. The call drops as soon as the fax answers.
Since I am using FreePBX
On 02/09/2012 01:17 PM, Danny Nicholas wrote:
If the MOH thing is really true, a more realistic test would be to run
playback(demo-instruct). Since I know that I will eventually cross this
bridge in real life/real time, I devised this test on my Asterisk 10.0 box
Dialplan (in default context)
Hi all,
I'd like to know how I can turn off the splash ring voicemail waiting
indication on a PAP2T from the provisioning XML file. I can do it from the web
interface, but I need to do it on a lot of machines
TIA,
--
Take care and have fun,
Mike Diehl.
--
The Asterisk Development Team has announced the release of Asterisk 1.8.9.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.9.2 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 10.1.2. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 10.1.2 resolves several issues reported by the
community and would have not been possible
Shaun,
snip
Just thinking out loud here but I'm guessing it may be fair to just
set alarmdebounce to 0 by default on gen1 cards.
With dahdi-linux 2.2.0.2 does your card function if you set alarm
debounce to 2500?
/etc/init.d/dahdi stop
sleep 3
modprobe dahdi
modprobe wct4xxp debug=1
nobody facing any issue with this or nobody using real time architecture
On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
Hi Group.
I am facing an issue with Peer registration in my asterisk server .
I am using asterisk version 1.8.5.0 and using SIP real-time
Thanks for reply and share your techniques, dialplans and knowledge on this
thread. But my question was not related to load-balancing. I want to know ,
Why freeSwitch can preferred with compare to Asterisk(Call base , quality
base)? And what is architecture difference between them.
I am totally
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