hello all,
i have problem in using g729 codec. my asterisk version is 1.8.22. when i
run core show codecs in asterisk, there is a g729 codec in the list so i
assume that i can use it for my channels. but connection can not be set
when i use it for my h323 channel.
i read somewhere that codec g729
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
s m sam.gh1...@gmail.com schrieb:
hello all,
i have problem in using g729 codec. my asterisk version is 1.8.22. when
i
run core show codecs in asterisk, there is a g729 codec in the list
so i
assume that i can use it for my channels. but
Hello;
We need to have admin page, so the administrator can create passwords to be
used to join the meetme conferences and can determine the allowed time ..
Well, the admin interface can be done easy (I do not know if there is something
ready), and the password and the time limitation can be
thanks Dominik,
you're right. i don't pay attention enough about my subject.
about g729, you mean if it get free g729 and all my systems (PBXs and
routers) use g729 codec for setting a call, call is set without any problem?
On Tue, Oct 1, 2013 at 10:35 AM, Dominik George n...@naturalnet.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Hi,
about g729, you mean if it get free g729 and all my systems (PBXs and
routers) use g729 codec for setting a call, call is set without any
problem?
Yes, if all systems use g729 directly, you are ready to go.
- -nik
-BEGIN PGP
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9
as if my PBX machine is trying to authenticate to itself. It seems
someone is attacking my asterisk PBX.
Is there a way to fix this problem?
Thank you.
Giorgio
Hi,
my asterisk server has a strange behavior when all trunks are
unreachable (for example due to internet connection), it doesn't accept
registrations from any internal extension, and, obviously, I can't do
any internal call.
On the log I see only lines like:
[2013-09-30 01:35:04] NOTICE[1827]
On Tuesday 01 October 2013, Francesco Namuri wrote:
Hi,
my asterisk server has a strange behavior when all trunks are
unreachable (for example due to internet connection), it doesn't accept
registrations from any internal extension, and, obviously, I can't do
any internal call.
Asterisk
Il 01/10/2013 10.16, A J Stiles ha scritto:
On Tuesday 01 October 2013, Francesco Namuri wrote:
Hi,
my asterisk server has a strange behavior when all trunks are
unreachable (for example due to internet connection), it doesn't accept
registrations from any internal extension, and, obviously,
Well, you could use some software like denyhosts or fail2ban to block an IP
after a predefined number of (failed) authentication attempts.
Regards,
Ricardo
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.comwrote:
Hi,
I get a lot of these messages on my Asterisk CLI:
Hi Ricardo,
we are already using fail2ban but it bans my own ip address not the real
original ip of the attacker. How can I find it?
Thank you
Giorgio
On 10/01/2013 02:16 PM, Ricardo Saffi Marques wrote:
Well, you could use some software like denyhosts or fail2ban to block
an IP after a
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com
wrote:
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user
Look at Web-MeetMe ( http://sf.net/projects/web-meetme )
If you are on Asterisk 1.6.7 or later you have access to RealTime
MeetMe conference storage, otherwise you need to use a
script and Asterisk application included with the WMM download.
Dan
From: asterisk-users-boun...@lists.digium.com
Hi,
Bad boys trying to guess a valid username.
in sip.conf uncomment alwaysauthreject=yes and Asterisk always reject 1st
invite.
On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM,
Hello,
With setvar statements in chan_dahdi.conf, we have a convenient way to
store DAHDI channels specific values.
Unfortunately, we don't have a function to access this data from the
dialplan as easily as SIPPEER ou IAXPEER would for SIP or IAX trunks.
Using AST_CONFIG, you can access DAHDI
15 matches
Mail list logo