initially muted).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Alessandro Russo
*Sent:* Thursday, December 11, 2008 2:42 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject
% 99.937401% 99.968460% 99.967667% 99.936333%
--- Results after 22 passes ---
Best: 99.972 -- Worst: 99.935 -- Average: 99.955330, Difference: 99.992836
Any suggestions?
Alessandro R.
On Fri, Dec 12, 2008 at 7:39 PM, Matthew J. Roth mr...@imminc.com wrote:
Alessandro Russo wrote:
we
This is because meetme needs zaptel to works:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe
Please note: A Zaptel timer must be present for conferencing to work! See
Asterisk
timer http://www.voip-info.org/wiki/view/Asterisk+timer
Alessandro R.
On Thu, Aug 23, 2007 at
Hi Asterisk Users,
we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323
1.18.
We are using MeetMe for conference calls and with two participants there is
no echo problems, but with more than two participants there is a lot of echo
that sometimes disappear for a short time
Hi to all,
I would like to limit the numbers of inbound h323 connections for different
extensions, for instance, I've the following rules in my dialplan:
exten = 123,1,DIAL(H323/1100)
exten = 234,1,DIAL(H323/2200)
and I would like to limit to 5 the number of h323 connections for exten 123
e
Hi to all
I'm using Asterisk 1.4 and I'd like to use LDAP instead of sip.conf...
Now you suggest to use asterisk realtime (res_config_ldap) or astirectory??
Can I use one of them with version 1.4?
thx
On 8/30/07, Gavin Henry [EMAIL PROTECTED] wrote:
No probs.
On 29/08/2007, Abhishek M S
Hi to all,
I've installed Asterisk 1.4 and all function very well.
Now I need to use LDAP or RADIUS instead of sip.conf since all the trusted
users have an account on LDAP/RADIUS.
Any suggestions...try astirectory (but is for asterisk 1.2.x, I've 1.4.9) or
Asterisk realtime LDAP (it is only for
Hi to all,
I'm using asterisk 1.4.9 with chan_h323.
When someone in the H323-VoIP cloud dial 1234 this number is assigned to my
asterisk-machine, so the VoiceGW forward the flow to my machine, asterisk
though the dialplan can delivery the call to a particular SIP phone...this
is ok...
I can also
Hi,
first step is correct
Hmm.. This is what I get:
[EMAIL PROTECTED] ~]# mysql -u root -p
Enter password:
Welcome to the MySQL monitor. Commands end with ; or \g.
Your MySQL connection id is 187143 to server version: 4.1.20
Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
on it, which makes use of the MySql
server as well (and all is ok there).
Adrian Marsh
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Alessandro Russo
*Sent:* 07 August 2007 14:13
*To:* Asterisk Users Mailing List - Non-Commercial
if using socket to connect .
On 07/08/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi, try to login as asteriskcdruser to mysql
# mysql -u asteriskcdruser -p
Enter password: password
Hi to all,
I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper.
I've tested h323 using ohphone and I can talk between them, then I've tested
SIP with Twinkle softphones and function very well.
Now I have to perform call from h323 to sip and viceversa.
How can I do it
I
.
Thx
On 8/6/07, map [EMAIL PROTECTED] wrote:
Hi Alex,
You should create a dial plan to route sip calls to H.323 calls.
Take a look at :
http://www.voip-info.org/wiki/
On 8/6/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi to all,
I've installed Asterisk 1.4.9 with h323, and gnugk
Hi,
I'm using H323 in asterisk 1.4.9
work well
On 8/3/07, yonoko molomo [EMAIL PROTECTED] wrote:
Hi,
I have used h323, oh323 and ooh323.
My experience is that ooh323 does not work properly, i dont recommend it.
I dont know why, but the sound is bad, with sound breaks. I also need
to put
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