Hi all,
I just completed an upgrade from 1.6.2 to 10.2.1 and I've run into a few
issues.
The issue I'm trying to fix rght now is that after the upgrade, I no longer get
any
CDR's. I'm trying to log to a Mysql database via odbc.
When I do an odbc show all, it shows that it's connected.
2012/8/9 Ashish Agarwal ashisha...@gmail.com
Hello,
I have two asterisk boxes running and both are using DAHDI PRI Card. I
wish to know if IAX is the best method to connect both the boxes?
IAX2 is a great protocol, it can do amazing things in saving bandwidth
(with the trunking feature) and
The IP Phones that I am using is the Digium D40, but did not find any place to
enable the RPID, I do not know if they are enabled by default.
Any advise?
Regards
Bilal
--
Hi,
You need to set rpid on the calling phone settings, if that
phone knows
what to
- Original Message -
From: Support mdi...@diehlnet.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 9, 2012 1:54:21 AM
Subject: [asterisk-users] No CDR after upgrade (1.6.x - 10.2.1)
However, when I complete a
On 8/9/2012 6:32 AM, bilal ghayyad wrote:
The IP Phones that I am using is the Digium D40, but did not find any place to
enable the RPID, I do not know if they are enabled by default.
Any advise?
Regards
Bilal
Remove the allowrpid=yes as this in incorrect.
In sip.conf for the extensions in
On 06/08/12 13:48, Daniel-Constantin Mierla wrote:
* http://asipto.com/u/68
The tutorial focuses on how to use Asterisk's database structure to
perform authentication in Kamailio SIP server, along with user location,
nat traversal, instant messaging, presence, a.s.o., offloading
- Original Message -
From: Mike Diehl mdi...@dominion.diehlnet.com
To: asterisk-users@lists.digium.com
Cc: Matthew Jordan mjor...@digium.com
Sent: Thursday, August 9, 2012 12:08:01 PM
Subject: Re: [asterisk-users] No CDR after upgrade (1.6.x - 10.2.1)
On Thursday 09 August 2012
Hi There,
Is should be possible to CONFGURE Asterisk as a multi-tenant IVR server
right? Like tweaking configuration to configure a multi-tenant PBX with
Asterisk.
Thanks.
--
_
-- Bandwidth and Colocation Provided by
On Thu, Aug 9, 2012 at 10:59 AM, Kannan vasdevelo...@gmail.com wrote:
Is should be possible to CONFGURE Asterisk as a multi-tenant IVR server
right? Like tweaking configuration to configure a multi-tenant PBX with
Asterisk.
I don't know why you make a distinction between a multi-tenant IVR
Hi,
I've successfully setup Asterisk on my local PC and can make call using
Twinkle to the server. But, I cannot call to my Asterisk server at
Rackspace. I have been trying several things to figure it out, no luck. My
PC is behind NAT, so I've set that up in sip.conf (nat=yes). I can ping my
Hi,
I have used basic Asterisk as a PBX controlling few extensions. My question is
simple, at work there is an existing Call Manager/PBX and all which
manages all the extensions for SCCP VOIP phones. Can Asterisk be used to
manage just 1 VOIP phone and still can make internal calls to the
This shouldnt be a problem. Asterisk is basically flavor-blind as to
what type and quantity of phones you put on it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge Díaz
Sent: Thursday, August 09, 2012 4:21 PM
To:
Danny, thanks for your input...
Can you tell me if I am wrong with the following or give me a brief guide
of what to look at?
I was planning on using Asterisk + chan_sccp to control the VOIP phone.
Asterisk will NOT replace the current CCM/PBX at work, it will have just
one phone but in a way
You will need to setup a SIP trunk between the asterisk server and the CCM
server. Then in your asterisk config, you'll need to direct any extensions that
are handled by the CCM server to that trunk. You'll also need to configure the
CCM server to send calls to the specific extension through
Hi,
Asterisk is quite good with resolving the NAT issues specially the kind of
issue you are facing ,as I see it, shouldn't be a problem. A few steps you
can troubleshoot this problem.
1-a:Are your SIP packets from PC/SoftPhone reaching the server !! On
Asterisk CLI execute *CLIsip set debug on
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