Hi,
I'm trying to find the differences between the two CLI gain parameters
of Dahdi : dahdi set swgain and dahdi set hwgain.
When I change one of these parameters the output of :
asterisk -rx dahdi show channel X | grep Gains
don't show me any changes.
Did dahdi show channel X shows HW or SW
Hi everyone,
It appears that in Asterisk version 1.8.10.1 happens the same problem:
asterisk1*CLI core show version
Asterisk 1.8.10.1~dfsg-1ubuntu1 built by buildd @ yellow on a x86_64
running Linux on 2012-04-24 12:47:04 UTC
The sip calls doesn't work after next log message:
[Aug 22 13:56:44]
Hello everyone,
Recently we've been focusing quite heavily on making Adhearsion[0] faster. To
do that, we needed a convenient way to test our Asterisk voice apps. The
obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a
little clumsy to use sometimes, especially if
On 27 Aug 2013, at 15:34, Ben Klang wrote:
But what's REALLY useful is Sippy Cup's ability to dynamically generate PCAP
audio. If you've ever needed to drive an IVR from SIPp you're probably
familiar with the pains - it usually requires capturing an actual call,
isolating the RTP, and
Hi, Guys.
I wanted to know from you, if you know: 1) any companies in Ireland that
develop software and hardware for Asterisk integration with legacy
interfaces (E1, T1, FXO, FXS...) . 2) any companies in Ireland that work
with Asterisk development (modules, channel drivers...).
I work with
Hi,
Maybe you can get in touch with them:
http://www.innovate.ie
- Laszlo
2013/8/27 Gustavo Meira grnme...@gmail.com
Hi, Guys.
I wanted to know from you, if you know: 1) any companies in Ireland that
develop software and hardware for Asterisk integration with legacy
interfaces (E1,
- Original Message -
From: Noah Engelberth nengelbe...@team-meta.net
I have an Asterisk 11.5 system, using SIP Realtime and operating as a
ITSP. One of my customer’s endpoints is a NetVanta 7100 PBX system
that has a SIP trunk connection to my Asterisk box. The NV 7100 has
a public
You have pretty much found what the issue is. The AdTran is not properly
incrementing the SDP version.
Michael
(elguero)
Thank you for your reply. I figured that was what was happening, but didn't
know of a good place to go looking so I could forward the right information on
to
Hey All,
Growing call center. Currently at about 200 call center staff, running
about 1000 calls per hour. Gearing up to double that. Not too sure that
a single server will support that growth. So, I'm trying to come up with
ways to scale the system and still maintain a simplistic design. So
Hi Greg,
I have a similar setup with multiple asterisk boxes. I have used opensips
as a load balancer (+some pre routing logic before load balancing).
It is based on this idea:
http://www.opensips.org/Documentation/Tutorials-LoadBalancing-1-9
- Laszlo
2013/8/27 Gregory Malsack
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with CAPI in my first
installation (1 BRI), and a hfc 4 port bri card with misdn on other.
The first
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with CAPI in my first
installation (1 BRI), and a hfc 4 port
Thanks Laszlo, but if I'm reading this correct, this unit is load
balancing based on traffic type, transcoding/voicemail/etc/... As all of
my traffic would be the same, I don't see that this would fit too well.
Additionally, my problem is more, how do I maintain queue stats across
the work
Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with
Hi,
Will Keypress option will work when am in the queue and hearing MoH?
Lets say a caller is waiting in queue and while he is hearing MoH, can he
key in some DTMF and go to some other queue? is that possible?
Regards
--
_
--
On Tue, 27 Aug 2013 21:28:36 +0200
Gergo Csibra csi...@gmail.com wrote:
Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN
(CAPI or misdn) channels? I've tryed everything
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security
releases
are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3,
10.12.3-digiumphones,
and 11.5.1.
These releases are available
Asterisk Project Security Advisory - AST-2013-004
Product Asterisk
Summary Remote Crash From Late Arriving SIP ACK With SDP
Nature of Advisory Remote Crash
Asterisk Project Security Advisory - AST-2013-005
ProductAsterisk
SummaryRemote Crash when Invalid SDP is sent in SIP Request
Nature of Advisory Remote Crash
Yes you can. Check the 'context' parameter in queues.conf. When caller
presses a single digit extension while waiting in a queue, (s)he'll be
taken out of queue to this context. Then you can send caller to different
queue from this context.
--Satish Barot
Ahmedabad, India.
+919978599700
On Wed,
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