SInce no one else has brought this up, just thought I'd let you know that it
is being worked on...
http://bugs.digium.com/view.php?id=8824
And it works - I am using it for months already.
Note that not all phones support it. Cisco and Policom supports it, while Snom
does not.
Hi,
I appreciate the help. I called the vendor of the card and they recommended
removing all of the PCI cards on the system (including the video card), and
moving the card to a new PCI slot.
I did all of them together, ran the system headless, and ssh'ed in remotely. It
worked! haha...
Before an IP Phones can be registered to an Asterisk server, the extension for
it must be configured in Asterisk. Usually, Asterisk adminintor must add the
extension by hand. Is there any library, API to do this by software???
For example, i want to develope a software that add new extensions
Dear all
I have asterisk server with 2 E1 port now i want to
redendecy for my server means one of server goes down automatically second goes
in active mode is it possible and how to switch E1 to second server ??
-
Choose the
ENUM and ISN
You may be interested to know that John Todd was kind enough to come
by at the last minute and give us a thorough grounding in ENUM and
expand our knowledge about http://Freenum.org where you should run,
not walk, to get yourself an ISN (ITAD Subscriber Number).
You can listen to or
On Sat, Sep 08, 2007 at 12:42:34AM -0700, G B wrote:
Hi,
I appreciate the help. I called the vendor of the card and they recommended
removing all of the PCI cards on the system (including the video card), and
moving the card to a new PCI slot.
I did all of them together, ran the
You can use asterisk realtime which can read sip config from database (
mysql/pgsql) . Your application can just write info to database and asterisk
will read it and make peers . You can also include a custom config file
within sip.conf and make your application write peer settings to that file
On Sat, Sep 08, 2007 at 12:58:44AM -0700, phananhvu wrote:
Before an IP Phones can be registered to an Asterisk server, the
extension for it must be configured in Asterisk. Usually, Asterisk
adminintor must add the extension by hand. Is there any library,
API to do this by software???
From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04 -0500 Subject:
Re: [asterisk-users] Musiconhold instead ringing On Friday 07 September 2007
07:02:01 pm wassim darwish wrote: Hi: When i get an incoming call, i
Hi,
Am Samstag, den 08.09.2007, 09:44 + schrieb wassim darwish:
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04
-0500 Subject: Re: [asterisk-users] Musiconhold instead ringing On
Friday 07 September
On Saturday 08 September 2007 04:52, satish patel wrote:
I have asterisk server with 2 E1 port now i want to
redendecy for my server means one of server goes down automatically second
goes in active mode is it possible and how to switch E1 to second server ??
From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Sat, 8 Sep 2007 14:48:18 +0200 Subject:
Re: [asterisk-users] Musiconhold instead ringing Hi, Am Samstag, den
08.09.2007, 09:44 + schrieb wassim darwish:
I would try
[your-coming-in-context]
exten = s,1, Dial(type/identifier, timeout, m(your_moh_class), URL)
Hoai-Anh Ngo-Vi
Biedenkopfer Weg 13
60489 Frankfurt am Main
Email:[EMAIL PROTECTED]
Telefon: +49 (0) 69 74 22 36 63
Mobil.:+49 (0) 179 66 29 520
-Ursprüngliche
hi:
thanks a lot for your suggestion. i have setup up an experimental
environment like yours,
and dundi works great. but it's not easy for us to archive this in
real world. we have other pbx(like alcatel) that need to co-work with
asterisk. so area code with each branch office seems easier to
Hi,
I noticed today, that there was a stale SIP call on my 1.2.24 A*k
server. One call (X-lite client) started yesterday into a meetme
conference. For some reason the call stayed established.
From network stats, I see transmit data but no receive (as obviously the
client went offline).
G B wrote:
Date: Fri, 7 Sep 2007 07:13:38 -0400
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] New Installed X100p
G B wrote:
Hi,
I just installed an X100p knockoff (OpenVox a100lp). I have seen my
errors in the mailing list but
Barton Fisher wrote:
I'm using version 1.2 and need a method to detect the number of
channels in use
from inside the dial plan. I'd like to count total channels
system-wide, but even better
if I can determine for a selected extension also. I've searched the
wiki, and don't see such
a
Alex Balashov wrote:
There are lots of these. They belong to a class of appliance known as a
media gateway.
http://www.voipsupply.com/product_info.php?products_id=1038
If you REALLY want to pay that kind of money for something that serves
this purpose for a single T1... well, we'd all
I have had Digium tech support tell me to do the same thing
Thanks,
Steve
G B wrote:
Hi,
I appreciate the help. I called the vendor of the card and they
recommended removing all of the PCI cards on the system (including the
video card), and moving the card to a new PCI slot.
I did
Thank you both of you Jaswinder Singh, Tzafrir Cohen.
Though your guides don't directly solve my main problem but it gave me the
light from the end of the tunnel. So now i can deal with my headache problem.
I have 2 choices:
(1) Use Realtime
(2) Inclue configure file
Thanks again to
since the udev not installed in by the sequence, that may not supported in
your distribution, use the correct version of udev for linux kernel version.
i got the same problem with another device, udev wont create the
device node automatically, if yours seems to be the same, this
approach may solve
Only had 4 licenses for the G729 codec. Never had any trouble with
these.
Michael
--Original Message Text---
From: Olivier
Date: Fri, 7 Sep 2007 08:34:45 +0200
Michael,
How many simultaneous calls could you get ?
Regards
2007/9/7, Michael Graves [EMAIL PROTECTED]: I run Astlinux on a
T5700.
Hi all,
Have just installed v1.4.11 of Asterisk, but I am trying to have it start at
boot but with no luck.
I have used the make config command but it doesn't start. Any help would be
apreciated, many thanks!
___
Sign up now for AstriCon 2007!
Hi all
what is the difference between
show channels
sip show channles
i see the difference in both
show channels show me 30 channels
sip show channels shows me 221 channels
any description on this
ram
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Sign up now for AstriCon 2007!
http://www.bogen.com/products/voip/index.html
On 9/5/07, Dave Fullerton [EMAIL PROTECTED] wrote:
Carlos Chavez wrote:
I have a customer that has two buildings that are connected with a
fiber link. We have a single Asterisk server to cover both buildings.
Now the customer went and
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