Hello,
How can I set a hard limit to the number of Local channels asterisk can
spawn?
--
Thanks.
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Greetings Ron-
Just wanted to give you a heads up about an alternative SCCP channel driver
available for Asterisk. Please see here:
http://freecode.com/projects/chan-sccp-b
I have no experience with it (nor SCCP in general) but just wanted to give you
an option in the event the included SCCP
On 06/14/2012 04:20 AM, [Digital^Dude] ® wrote:
How can I set a hard limit to the number of Local channels asterisk can
spawn?
chan_local does not have a mechanism to do this.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP:
http://lists.digium.com/pipermail/asterisk-users/2012-February/270427.html
That worked for me with the polycom 3.x firmware; I haven't tried it with 4.0
firmware yet.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David
Hello,
Asterisk under 90% load of SS7 calls can only withstand the voice
broadcasting for 30 minutes. After around 30 minutes, it stops receiving
any call hits via AMI. No errors are reported. Giving it a minute's rest
makes it work for another 30 minutes.
Can anyone hint to what may be
I have an Asterisk (v10.2.0) running and bound to address ::. I think
this way he listens and answers to requests send to the IPv4 and IPv6
address (haven't check that with IPv6 yet). What I want to achieve is,
that he handles signaling via IPv4, but RTP via IPv6.
In my setup, I have a user
- Original Message -
From: Pawel Kuzak pawel.ku...@1und1.de
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, June 14, 2012 10:22:21 AM
Subject: [asterisk-users] Dualstack
I have an Asterisk (v10.2.0) running and bound to
Hello,
Thanks you for the replies ill take a look at the driver you sent over. Im
going to run some test and see what happens, hopefully the driver in 1.8 is
soild and nothing needs to be messed with, but we will see :)
On Thu, Jun 14, 2012 at 5:06 AM, Tim Nelson tnel...@rockbochs.com wrote:
That's my question...the sbc provides security over trunking, right? The
same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of
add-value to an Asterisk deployment?
A PBX provides functionality to users. An SBC *can* secure a PBX
against the outside world, but that is
Is the chan-sccp-b project the same one that got put in SVN of 1.8 branch?
I have not been able to find anything definitive that says so, I really
need 1.8 branch so trying to see which is the best way to go.
Thanks
On Thu, Jun 14, 2012 at 9:34 AM, Ron McCarthy ronmc...@gmail.com wrote:
Hello,
The Asterisk Development Team has announced a security release for Asterisk 10.
This security release is released as version 10.5.1.
The release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of Asterisk 10.5.1 resolves the
Asterisk Project Security Advisory - AST-2012-009
Product Asterisk
Summary Skinny Channel Driver Remote Crash Vulnerability
Nature of Advisory Denial of Service
We couldn't see anything about this on the Digium site, but maybe
someone here can comment?
Do the new Digium phones provide good teleworker functionality?
The benchmark we're comparing against is the capabilities of Mitel
3300 IP systems with Mitel 5330 IP phones (running their proprietary
On 06/14/2012 04:57 PM, asterisk users wrote:
We couldn't see anything about this on the Digium site, but maybe
someone here can comment?
Do the new Digium phones provide good teleworker functionality?
Yes, I believe they do :-)
The benchmark we're comparing against is the capabilities of
On Thu, Jun 14, 2012 at 4:05 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 06/14/2012 04:57 PM, asterisk users wrote:
We couldn't see anything about this on the Digium site, but maybe
someone here can comment?
Do the new Digium phones provide good teleworker functionality?
Yes, I
On Thu, 2012-06-14 at 16:23 -0600, asterisk users wrote:
This is pretty good news, overall. To comment on Kevin's points:
- The end-to-end encryption is important to us, because
client-ID-sensitive information is part of our environment. Something
like built-in OpenVPN would work for us,
On Thu, 14 Jun 2012 00:46:25 +
Klaverstyn, David C david.klavers...@intergraph.com wrote:
I have a Polycom Handset on a front door and I'd like the phone to
dial a number as soon as the handset is lifted without having to
press and buttons or enter any numbers. I know how to do this on a
Hi all, I have a project for the 3G related, AMR and AMR-WB support.
I'm using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.
Now I have to setup a SIP server/SIP PBX in our Lab for test, does the
Asterisk
support these
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