Hey Ron,
Did you get your ultramonkey setup working correctly?
I'm about to roll ultramonkey here, any tips?
Regards,
Igor H.
Nhadie wrote:
hi,
has anyone implemented ultramonkey with asterisk? do i really need to
setup fwmark as discussed in the url below? thanks!
Hi guys,
Does the Aastra line of phones work with dns srv records?
I'm trying to get my 8133i to do this and in the settings it asks for ip
addresses of registration and proxy servers.
Does this mean that it will not just let me put the domain name in like
other devices I have and then do fail
2008/10/3 satish patel [EMAIL PROTECTED]
2008/10/3 Joe Pukepail [EMAIL PROTECTED]
I think this is what you want: http://bugs.digium.com/view.php?id=8824
Thanks : this one very interesting.
Bottom line is it doesn't work at the moment right ?
http://bugs.digium.com/view.php?id=8824
Replying to myself, I've just read in 1.6.1 announcement that a new
Incomplete dialplan application is the one that provides what I'm looking
for ...
2008/10/3 Olivier [EMAIL PROTECTED]
Hi,
If my memory serves me right, there was thread (in dev mailing list ?)
explaining how we could
Hello.
Go to Zaptel dir and type
make uninstall
make uninstall all
make remove
Before removing Zaptel, be sure Zaptel is stopped.
/etc/init.d/zaptel stop
There are some files which not removed by make.
If necessary, you can delete these files manually.
But if Zaptel is not loaded, it's not
Hello.
Have you ever tried updating your GCC version?
Thanks.
On Thu, Oct 2, 2008 at 8:30 PM, Satish Patel [EMAIL PROTECTED] wrote:
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Thu, Oct 02, 2008 at 11:33:01AM -0400, Satish Patel wrote:
Regards,
Satish Patel
Hi,
You can see here and there, several new SIP RFCs relying on SIP Events
Framework.
For example, RFC3680 with which a registration server would notify endpoints
with relevant events.
In Asterisk 1.6.1, a new SIPnotify AMI command implements a mechanism to
send arbitrary NOTIFY commands.
Is
You realy have issues .. instead of wasting my time and the group's time and
your own time with such emails.. just ignor my emails from now on.. i've been
in this list for years now.. you are the first one who spoke of this .. and you
want me to change my email address?? how smart is that? live
On Sat, Oct 04, 2008 at 02:02:48PM +0200, Olivier wrote:
Hi,
You can see here and there, several new SIP RFCs relying on SIP Events
Framework.
For example, RFC3680 with which a registration server would notify endpoints
with relevant events.
In Asterisk 1.6.1, a new SIPnotify AMI command
Yup. Did that in the same setting flow, yet it didn't show up when TF3D was
off.
Oh well.
Someone said use Fring and I think so far it's worked over EVDO. Nice!
Thanks..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008
How can I prevent a remote DoS as described on the following site? :
http://www.voip0day.com/news/remote-denial-of-service-exploit-effects-the-asterisk-pbx/
Best regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262
I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup
Steve,
Steve Totaro wrote:
BT3 (BackTrack) LiveCD is one of the best things out there, even has
sipp built right in, as well as other great apps, utilities, and
security auditing.
I suggest everyone have a copy in their arsenal, and it is free of course.
What does it do? I mean, for
Hi,
I am using asterisk-1.4.11. Voicemail quotas only apply to the new messages in
the INBOX. Browsing quickly through the 1.6 app_voicemail it seems that 1.6
does implement voicemail quota for both INBOX and Old messages. Is that
correct? If so, is there an existing patch available that
This seems to be related to inbound calls. So would this work for music on
transfers within that context as well as hitting the hold key on calls?
--- On Fri, 26/9/08, Darrick Hartman [EMAIL PROTECTED] wrote:
From: Darrick Hartman [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Music on hold
Hi all,
I've googled around for concrete solutions on load balancing Asterisk, and
it appears there are several ways to skin this cat -- but not one solution
which is all appealing. I have the following requirements, which aren't
anything extraordinary:
* I need to handle roughly 300
One other thing you could try would be to use OpenSIPS and use a
standard config that routes to a hostname (with a creative failure
route setup). You'd then setup the hostname in DNS as multiple SRV
records reflecting your pool of Asterisk servers (set your TTL very
low for these records).
OpenSIPS/Kamailio have modules designed specifically for that kind of
functionality now without a need for an outside monitoring process or
SRV reliance.
Darren Sessions wrote:
One other thing you could try would be to use OpenSIPS and use a
standard config that routes to a hostname (with a
I know. :)
I've already mentioned some of the OpenSIPS options to him on the
OpenSIPS users list (LCR module specifically). Just brain dumping
everything that came to mind.
- D
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
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