I have been doing a lot of reading forums and elsewhere but am somehow
unable to connect the dots.
Here is what I am trying to accomplish initially and then wish for it to
grow bigger from here on.
I have two POTS (Analog) line that would connect to the Asterisk Box.
I have, to begin with 5 IP
Although it was better to ask it in Asterisk commercial list but you have
different options like Digium, Sangoma or Openvox. TDM410P is the PCI one
from Digium which suits your description. Just remember to buy two trunk
(FXO) modules too and if you are looking for a best sound qulity get
hardware
Hello,
I have done yum install asterisk18 freepbx and it has installed Asterisk
and FreePBX just fine. However, none of the CDR get recorded in
asteriskcdrdb table in MySQL. They are available
in /var/log/asterisk/cdr-csv/Master.csv. What configuration file sets the
setting for writing these CDRs
Poor grammar on my part,
What I meant was to assign one ext to each IP Phone, my initial setup
consists of 5 phones. If all things work out as planned and after better
understanding I wish to support upto around 70 IP phones.
I do plan to get echo canceller too ;)
Thanks for the reply.
On
Not sure about yum installs but in 1.8 I have had to move to using odbc as the
method to populate the mysql database
http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc
Cheers Duncan
On 17/06/2012, at 4:22 AM, Bruce B wrote:
Hello,
I have done yum install asterisk18 freepbx and it has
Did you install the addons
Yum install asterisk18-addons-mysql
-Original Message-
From: Duncan Turnbull dun...@e-simple.co.nz
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 17 Jun 2012 08:30:00
To: Asterisk Users Mailing List - Non-Commercial
Hello,
I have an internal extension, e.g. 1005 which is being called from an
external/public number like 123456789. Now when it comes to the spoken
voicemail information it says something like number 1000 not available,
however it should say number 123456789 not available. How can I configure
Amit,
Make sure you have an option to return Digium TDM410P if it does not
work for you.
In my experience Digium TDM410P produce substantial background noise on
certain Dell computers. Generic TDM400 do not have this issue.
On top of that FXO channels exhibit intermittent problems with
In my experience when you have intermittent problems with incoming caller ID,
FXS -- with DTMF detection you have to adjust your rxgain and/or txgain. I am
NOT a fan of Digium cards, but these CallerID and DTMF issues are simple and
solvable and not related to the card itself.
-Original
On Sat, Jun 16, 2012 at 04:37:06PM -0500, Vladimir Mikhelson wrote:
On top of that FXO channels exhibit intermittent problems with incoming
caller ID, FXS -- with DTMF detection. These two problems manifest
themselves with both Digium and generic cards. It looks like it is just
a
Eric,
I wish it were that simple as you described.
rxgain, txgain, cidrxgain, Digium support, Jira ticket, hours of
troubleshooting, hardware swaps, external power source, etc. -- all
these are the options I tried so far.
BTW, I did state that these specific issues are not limited to Digium
On 6/16/2012 5:38 PM, Shaun Ruffell wrote:
On Sat, Jun 16, 2012 at 04:37:06PM -0500, Vladimir Mikhelson wrote:
On top of that FXO channels exhibit intermittent problems with incoming
caller ID, FXS -- with DTMF detection. These two problems manifest
themselves with both Digium and generic
Eric,
I wish it were that simple as you described.
rxgain, txgain, cidrxgain, Digium support, Jira ticket, hours of
troubleshooting, hardware swaps, external power source, etc. -- all
these are the options I tried so far.
BTW, I did state that these specific issues are not limited to Digium
On 6/16/2012 5:38 PM, Shaun Ruffell wrote:
On Sat, Jun 16, 2012 at 04:37:06PM -0500, Vladimir Mikhelson wrote:
On top of that FXO channels exhibit intermittent problems with incoming
caller ID, FXS -- with DTMF detection. These two problems manifest
themselves with both Digium and generic
I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi to
increase the duration of transmitted DTMF on your DAHDI channels. If that
fixes it, try lowering it. I find 80 usually works with even the worst IVRs.
-Original Message-
From:
Eric,
Thank you for the suggestion.
In fact the problem is with FSX channel which fails to catch some DTMF
tones from a phone which places an outgoing call. Shaun's theory was a
delay related to swapping.
-Vladimir
On 6/16/2012 7:40 PM, Eric Wieling wrote:
I was assuming incoming DTMF
You have verified this by using the Asterisk's DTMF debug option?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Saturday, June 16, 2012 9:37 PM
To: Asterisk Users Mailing List -
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