Hi all,
I'm trying to setup a system such that when a call comes in to Asterisk, it
first checks the account balance of the caller via Radius and then determine
if the call should go through or not.
I have an average experience in Asterisk but I'm quite new to Radius so I'm
not sure if this
Hi amit,
Thanks for the quick reply.
I'll look into this and hopefully get this to work. Thanks again!
Regards,
Ronald
On Wed, Sep 21, 2011 at 2:45 PM, amit anand onewaytoconn...@gmail.comwrote:
Hi
for this you need to write some agi script that will handle the other
feature.
Also you
Hi,
I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone
behind a different NAT network.
Asterisk - Nat - Internet - Nat - Softphone.
I can register my softphone to the asterisk box ok via SIP but the RTP
stream from the asterisk box is addressed to the private
Hi Tamer,
Many thanks for your comments, really your comments are useful. And finally
I think using dahdi instead of mISDN is better.
On Wed, Sep 21, 2011 at 3:10 AM, Tamer Higazi th9...@googlemail.com wrote:
Am 20.09.2011 19:47, schrieb Gopal krishnan:
What is the difference between using
Be sure, if you make us of HFC Boards that you have the zapfhfc patches.
There is some work for you to accomplish, like patching dahdi to make
use with the cheap isdn boards.
For office using ISDN Devices it's fairly enough. If you want to make
use of a server, I advise you to take the digium or
Execute a shell script instead -- too bad they have a small limit, but
that should work.
Ikka - Mitra Kreasindo ikka.vert...@mitrakreasindo.com wrote:
Is anyone can help me with this ? I'm really desperate.
Thx in ad.
From: asterisk-users-boun...@lists.digium.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Tuesday, September 20, 2011 7:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fixing an old bug related to extension s -
(Top posting 'cause that's what others did--and I like it that way, anyway.)
Not so obvious that a single mic can't be shared--if one call is muted, it
would work great.
This split-ear feature would be handy when, while on hold/in queue on call
A, you want to answer call B. You want to talk to
Hello,
Is Dahdi 2.5.0 supposed to support BRI NT PtmP mode ?
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On Wed, Sep 21, 2011 at 04:13:50PM +0200, Olivier wrote:
Hello,
Is Dahdi 2.5.0 supposed to support BRI NT PtmP mode ?
Not really. It is mostly out of its scope. Look into e.g. Asterisk =
1.8
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
2.4.x does it with me, so I am sure 2.5.x do makes it either!
Tamer
Am 21.09.2011 16:13, schrieb Olivier:
Hello,
Is Dahdi 2.5.0 supposed to support BRI NT PtmP mode ?
Regards
--
_
-- Bandwidth and Colocation
So the bottom is :
NT PtmP works with Dahdi 2.5 and Asterisk 1.8 !
Though I can't check this by myself at the moment, I'm adding a SOLVED note
in this reply to let others be aware, just in case.
Thanks for replying !
--
_
--
Just want to see if anyone else has seen something like this.
Setup:
Dell PowerEdge 2950
2 x Sangoma A104-D
Asterisk 1.8.5.0
Asterisk stop sending calls to agents and nothing stuck out in the
messages log, but at the same time I did see this error in
/var/log/messages
snmpd[1882] general
Ok Thank you Tamer.
On Wed, Sep 21, 2011 at 4:32 PM, Tamer Higazi th9...@googlemail.com wrote:
Be sure, if you make us of HFC Boards that you have the zapfhfc patches.
There is some work for you to accomplish, like patching dahdi to make
use with the cheap isdn boards.
For office using ISDN
Hello everyone.
Is there any way to get SIP Call-ID for a B-leg if call was not bridge?
For example when we get 480 or 486 SIP response or any other error code?
It seems that for this case BRIDGEPVTCALLID variable is not being set in
ver. 1.6.2.20 and 1.8.6.
Best Regards
Krzysztof
I am looking for a simple way to send occasional faxes via the FXO
port on my SPA3102 -- without having to connect a fax modem to an
ATA. In an ideal world, this would be some sort of softfax that
runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with
T.38.
This is one of those
On 22/09/2011 4:12 AM, Ian Pilcher wrote:
I am looking for a simple way to send occasional faxes via the FXO
port on my SPA3102 -- without having to connect a fax modem to an
ATA. In an ideal world, this would be some sort of softfax that
runs on my Linux desktop and talks (via Asterisk) to the
You can use ictfax HTTP://www.ictfax.org web interface to send faxes, Ictfax
is pure foip software based on t.38 as compared to hylafax
No need for iaxmodem and client application
On 22-Sep-2011 4:00 AM, Larry Moore lmo...@starwon.com.au wrote:
On 22/09/2011 4:12 AM, Ian Pilcher wrote:
I am
Hi,
Is it possible to change the default voice prompt for Asterisk meet me
conference bridge. We have our own customized recordings for Welcome and PIN
request and would like to use that instead of the default Please enter
your.. .
If I replace the default sound file with my custom file by
UmmmWhen I was a child I replaced the prompts to do that, Now I'd
suggest you to try creating a new directory in /sounds folder like /en i.e
/meetme and put in corresponding prompts there.
Then just before going into the meetme application change the Language for
the current call in dial plan
20 matches
Mail list logo
