You could base your box on a motherboard with an onboard CPU (like Intel Atom). The disadvantage
of these boards is that they usually come only with a single PCI or PCIe slot. There are
industrial boards with different options, but they are rather expensive.
The idle power of Sandy/Ivy Bridge
On 12 July 2013 16:36, Richard Mudgett rmudg...@digium.com wrote:
On Fri, Jul 12, 2013 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm using asterisk 1.8 on CentOS 5
I'm initiating call recordings with MixMonitor and trying to pause them
with the features.conf.
Whenever I
On 15/7/13 3:00 am, bilal ghayyad wrote:
I need to be able to send SMS messages for campaign or for specific users, also
I need to be able to receive SMS messages and do automatic reply.
In my experience, SMS is something best done out of Asterisk. That's not
to say that Asterisk can't do
Hello All,
I'm planning to use Asterisk only for voicemail Application and Recording
will be done at different server.
When user changing his personal greeting or leaving voicemail Call need to
throw to external Voicemnail recording server over SIP til the time
recording complete.
While
Hi,
1. Java process sends a request (e.g., add member to queue)
Do you see the TCP ACK coming back from Asterisk?
Yes, I do.
During the quiet period while you're waiting for the response, do you
receive events over that AMI connection?
Yes.
Are there other actions that you're
What would be a reasonable delay time? In the case I'm looking at right
now, the longest I can see is 7.2s.
Looking in the Java app logs, I can see it occasionally (166 times over
the last two weeks) timing out after five retries, which means it failed
to get a response to any of the
When you have many calls, there are usually (read/write=all) a lot of RTP, RTCP, and VarSet
events. This might slow down things, but whether they occur or not depends on your configuration.
This might be another thing to look at.
jg
--
On Mon, Jul 15, 2013 at 7:59 AM, jg webaccou...@jgoettgens.de wrote:
When you have many calls, there are usually (read/write=all) a lot of RTP,
RTCP, and VarSet events. This might slow down things, but whether they
occur or not depends on your configuration.
This might be another thing to
I need to make a Asterisk 18.0's offline compiling, SVN mp3 support
sources downloading does't particulary works cause my asterisk is in an
isolated network with NO network access whatsoever, I ve read this thread (
http://lists.digium.com/pipermail/asterisk-users/2013-June/279298.html) but
I 'm
Hi
You must copy the directory mp3, to the addons directory, where you put the
source asterisk code, and recompile it, again.
Kind Regards
On Mon, Jul 15, 2013 at 9:25 AM, leonardo collantes leonardo07...@gmail.com
wrote:
I need to make a Asterisk 18.0's offline compiling, SVN mp3
I guess this was a question for Alexander. As far as I am concerned, I never had such a load
that slowed down AMI event processing (responses within at most 1/10 of a second), but for
future tests I should probably set up a real torture test.
For a robust PBX application, it would make sense
Hi,
I am using asterisk 1.8.22 and have a problem when calling in parallel
several SIP endpoints and I am not sure how to resolve it. In this case
Asterisk will not bridge any audio to the caller before the 200 OK. Which
means any progress announcements, including remotely generated ringback,
are
On 15 July 2013 15:14, Hristo Trendev dist.li...@gmail.com wrote:
Hi,
I am using asterisk 1.8.22 and have a problem when calling in parallel
several SIP endpoints and I am not sure how to resolve it. In this case
Asterisk will not bridge any audio to the caller before the 200 OK. Which
means
On Monday 15 July 2013, leonardo collantes wrote:
I need to make a Asterisk 18.0's offline compiling, SVN mp3 support
sources downloading does't particulary works cause my asterisk is in an
isolated network with NO network access whatsoever, I ve read this thread (
On Sun, July 14, 2013 18:36, bilal ghayyad wrote:
Hello;
Anyone used PoE L2 network switches other than cisco and recommend
this for us? We need it to be stable and costly effective.
Regards
Bilal
We use multiple Cisco SG100D-08P eight-port POE unmanaged switches
with each located close
Unless it runs IOS, I don't think most of us would consider that box a Cisco
Likely it is a Cisco branded switch with Linksys hardware, i.e. consumer grade
stuff.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Yes, that is one of the former Linksys branded switches, now labeled as
Cisco Small Business.
I've used quite a few in small business installs. They work well, are
unmanaged, and inexpensive.
Jerome
On Mon, Jul 15, 2013 at 10:40 AM, Eric Wieling ewiel...@nyigc.com wrote:
Unless it runs IOS, I
On Mon, Jul 15, 2013 at 8:40 AM, Eric Wieling ewiel...@nyigc.com wrote:
Unless it runs IOS, I don't think most of us would consider that box a
Cisco Likely it is a Cisco branded switch with Linksys hardware, i.e.
consumer grade stuff.
They work well in small business. They have a command
I have a few TP-LINK TL-SF1008P and D-Link DGS-1008P running in office environments, but I
prefer the D-Link DGS-1210-10P (with fan) at a central location if the cable lengths permit it.
A couple of years ago I had 2 broken Netgear devices that ran about half a year, but I cannot
say anything
Something to check out:
http://www.kickstarter.com/projects/smush/smart-sms-texting-for-everyone-the-smushbox
I'm not affiliated with them at all, but have done business with the
company on other things and have always been happy.
On Mon, Jul 15, 2013 at 1:57 AM, Chris Bagnall
On Monday 15 July 2013, bilal ghayyad wrote:
Hello;
I need to be able to send SMS messages for campaign or for specific users,
also I need to be able to receive SMS messages and do automatic reply. Do
I have to use dongle or extra channel? What is the difference?
Also, I read that there is
I think I have found the answer to my questions in the source code of Dial:
case AST_CONTROL_PROGRESS:
ast_verb(3, %s is making progress passing it to %s\n,
ast_channel_name(c), ast_channel_name(in));
/* Setup early media if appropriate */
if (single !caller_entertained
The Asterisk Development Team has announced the release of Asterisk 1.8.23.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.23.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 11.5.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.5.0 resolves several issues reported by the
community and would have not been possible
How does one do this? We have a particular SIP phone that needs a large
jitterbuffer, but all I can see is how to put it on the *read* side of
the channel.
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If you want to store in external, why can't you have a NAS device and mount
to Asterisk server, let the mounted be a part in asterisk.conf, so that
voicemail will get recorded in external server...
Will it makes sense... !
Thanks.
On Mon, Jul 15, 2013 at 4:19 PM, Amit Salunkhe
On Jul 15, 2013, at 3:35 PM, Richard Kenner ken...@gnat.com wrote:
How does one do this? We have a particular SIP phone that needs a large
jitterbuffer, but all I can see is how to put it on the *read* side of
the channel.
At the risk of being a little tangential, what is a write-side
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Behrens
Sent: Monday, July 15, 2013 6:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Jitter buffer on write
Hi All,
I'm hoping someone can recommend a method to integrate Microsoft CRM with
Asterisk. Preferably an open source product otherwise a commercial product.
Regards
David Klaverstyn
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