ets from unknown that are replied with
Forbidden.
It's possible to use more than one trunk to the same provider?
If yes can someone help me?
Regards
Luis Silva
--
_
-- Bandwidth and Colocation Provided by http://
version is, 1.2?... But is what is happening to me. I'm
putting bind in the asterisk server and make some tests.
The sip modules blocks in dns queries but if doesn't block in sip
registrations timeout that works for me...
Regards
Luis Silva
>2010/3/26 Luis Silva
>
>> Hi a
Hi again,
In other asterisk it happened the same... No internet, no justvoip
resolution, no sip...
Remove the trunk, sip up... I'm going to test using bind with a "local"
zone.
More ideas/suggestions?
Regards
Luis Silva
>Hi ,
>
>I had some problems in the past with si
unreachable)
This is supposed to be like this? There is no dns tunning for asterisk+sip?
To avoid this I'm starting to think putting bind in the asterisk server and
publishing there the zones of the sip trunks. (Or instead of names start
using the ip's)
Any
Hi Philipe
>
>Hi!
>> I started to have problems with sip trunks, using more than one trunk
>> (and sometimes using only one) the sip module seems to freeze... My local
>> extensions lost registration and also the trunks. The only way that I
>> can restart the sip is removing the trunks...
>Have
Hi,
I need some help debugging a sip situation.
I started to have problems with sip trunks, using more than one trunk (and
sometimes using only one) the sip module seems to freeze.
My local extensions lost registration and also the trunks. The only way
that I can restart the sip is removing the
;Kevin P. Fleming wrote:
>> Luis Silva wrote:
>>> I have an asterisk in 1.2 version with 30 g729 licenses. I what to
>>> upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.
>>>
>>> For what I understand I can make the backup
Kevin P. Fleming wrote:
> Luis Silva wrote:
>> I have an asterisk in 1.2 version with 30 g729 licenses. I what to
>> upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.
>>
>> For what I understand I can make the backup of the license
operating system,
but in this case with new O.S and new version can I reuse this licenses?
Regards
Luis Silva
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http
Hi,
Still I can manage to have good incoming calls from h323. Can someone give
me a hand?
Regards,
LS
Date: Thu, 16 Jul 2009 15:46:43 +0100
From: Luis Silva
Subject: [asterisk-users] H323 situation
To:
Message-ID: <00ab01ca0624$3c9f69b0$b5de3d...@si
Hi all,
I have this installation:
Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0.
I have a problem that is, when a call comes from H323 and goes to a Sip
phone the asterisk sends two rtp streams to the sip. I checked this with
tcpdump, save the payload (voice is in G71
11 matches
Mail list logo
