Happy day to ya list!
I've recently deployed an update on our Asterisk server, taking it from 1.2
up to 1.8.5 - going from zaptel to dahdi. Excitement levels are high and
performance so far is wonderful.
The issue I have is I've purchased several G729 licenses, registered them,
installed the
Give zttool a try...
On Mon, Sep 26, 2011 at 2:32 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
thanks for your response but there is no dahdi becouse i have asterisk 1.4
installed (zaptel)
any help please
2011/9/26 Danny Nicholas da...@debsinc.com
Or
/bin/llsmod|grep
Greetings List!
I'm currently rolling out a new deployment of Asterisk 1.8 to replace
existing 1.2 servers...and have run into an issue which could use your
assistance!
For testing I have trunked (iax2) two of the servers - one running 1.8 and
the other at 1.2. Calls placed from SIP -- SIP
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *JT
*Sent:* Thursday, September 08, 2011 1:52 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Jitter only affecting meetme - and echo
testing
** **
Greetings List!
** **
I'm
.
JT
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Rosedale
Sent: Wednesday, July 06, 2011 5:37 PM
To: jonathan.tho...@us.patersons.net
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users
Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat
of a band-aid to the issue. But in my observations there is one clear
indicator that I am shocked is not used.
When I have done this test - pulling the network cable on a device during a
call - Asterisk actually reports
Good day all!
I have an issue which has plagued me for quite sometime now...and as I close
in on its cause, I have reached a point where additional info would be
greatly helpful!
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one
Hey List-ee's!
We're on Asterisk 1.2 with a PRI/T1 and maybe 50 phones.
Issue: When a conference is ongoing (meetme), if a call is placed to a PRI #
(CALLER A) the audio from the conference is heard by that caller while the #
they dialed is ringing in the background. The opposite also occurs
If all signs point to mis-configuration of your firewall, why not prove them
wrong (while in the process getting more details) just add wireshark to
the mix. You can then watch the traffic and be able to quickly identify if
any is being lost due to blocked ingress/egress ports.
DJ
On Sat,
to determine if Asterisk is doing this or if it's occurring outside
of my control.
-JT
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Will ANI delivery be required for Toll-Free service? (I'm assuming Yes
if we want to pass our caller id?)
Thanks a ton for your time,
JT
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Will ANI delivery be required for Toll-Free service? (I'm assuming Yes
if we want to pass our caller id?)
Thanks a ton for your time,
JT
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To UNSUBSCRIBE or update options
I've google for hours trying to find a discussion of a similar problem as the
one I'm having, so forgive me if this has come up before. If it has, please
point me in the right direction!
The problem occurs when a caller (A) is transferred by an intermediary party
(B) to voicemail (Voicemail or
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