I agree...a php/forum based solution like for example
http://www.woltlab.info/en/produkte.php would be more effective and easier
to manage.
Ricardo Villa
http://www.telesip.net
- Original Message -
From: James Taylor [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, August 09, 2003
-Term with the same command I can't seem to
reproduce it. Could it be an environment variable from the Terminal? I
looked but nothing seemed obvious.
Regards,
Ricardo Villa
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 05, 2003 4:25 PM
On Tue, 5 Aug 2003, Ricardo Villa wrote:
Hi,
Whenever someone leaves a Voicemail in our system we get this message on
the
console:
NOTICE[18447]: File sched.c, Line 209 (sched_settime): Request to
schedule
in the past?!?!
Does anybody know what it means?
Ricardo
I am testing a with 533Mhz Celeron/ 256MB. I guess this is certainly low
end.
Thanks,
Ricardo
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 06, 2003 3:34 AM
Subject: Re: [Asterisk-Users] Wierd Message
Ricardo Villa wrote
Hi Todd,
This limit on outbound calls looks interesting. Can you provide an example?
I have not used db routines before.
Thanks,
Ricardo Villa
http://www.telesip.net
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 07, 2003 2:51 PM
Hi,
Whenever someone leaves a Voicemail in our system we get this message on the
console:
NOTICE[18447]: File sched.c, Line 209 (sched_settime): Request to schedule
in the past?!?!
Does anybody know what it means?
Ricardo.
___
Asterisk-Users mailing
license usagemaui*CLI
There are currently 2 licenses of G.729 codec in use
Ricardo Villa
http://www.telesip.net
Hi Andrew,
After looking at some SIP messages again I too think the (c) field in the
SDP is what determines the RTP endpoints. It's just that in our case it is
always the same as the Contact field. In any case what you see here is
that * is making some changes here to make sure SIP messages
to adjust it between 10-60ms.
Thanks,
Ricardo Villa
http://www.telesip.net
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 28, 2003 2:04 AM
Subject: Re: [Asterisk-Users] g729 Codec
Its just like any other codec so it should work in SIP, IAX
h all Digium
cards."
Can somebody tell me please?
Thanks,
Ricardo Villa
http://www.telesip.net
it works perfect.
BR,
Dan
- Original Message -
From: Ricardo Villa [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 26, 2003 5:47 PM
Subject: [Asterisk-Users] PCM Voice Quality Issue on CVS Version
Hi,
I have asterisk-0.4.0 running. When I make a call between
e something that can be fixed on the CVS
version to prevent this problem?
Thanks,
Ricardo VIlla
http://www.telesip.net
12 matches
Mail list logo
