a remote X-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 PROTECTE
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]>
Sen
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] Wie
ce it's way in the past ...
>
> regards
> Martin
>
> 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, L
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
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
*CLI> sip show
channelsPeer
User/ANR Call ID Seq
(Tx/Rx) Lag Jitter Format0 active
SIP channel(s)maui*CLI> g.729 show 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 and
Right...The config file itself is encrypted so if you capture the downlaod
and can crack the RC4 algorithm then you are in.
The SIP authentication itself is just an MD5 hash of the password. If it is
a short password you can try to brute force your way into cracking it. But
if it is a long one (
The authentication goes out as an MD5 hash I think.
You could try to crack it:)
Ricardo
http://www.telesip.net
- Original Message -
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Ricardo Villa" <[EMAIL PROTECTED]>
> On Thu, 2003-07-31 at 10:07, Ricardo Villa wrote:
> > There is no way for you to know the vonage password associated with your
> > account. Even if you sniff out the tftp download, its encrypted.
>
> Is there any comparable service that isn't as anal? Or even better, is
&
There is no way for you to know the vonage password associated with your
account. Even if you sniff out the tftp download, its encrypted.
Ricardo
http://www.telesip.net
- Original Message -
From: "Steve Meyers" <[EMAIL PROTECTED]>
To: "Asterisk List" <[EMAIL PROTECTED]>
Sent: Thursday,
I was quoted $75 and $85 USD today.
Ricardo Villa
http://www.telesip.net
- Original Message -
From: "Joe Cooke" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 6:31 PM
Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 & 102
turns out.
Thanks,
Ricardo Villa
http://www.telesip.net
- Original Message -
From: "Dan Fernandez" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 28, 2003 5:08 PM
Subject: Re: [Asterisk-Users] g729 Codec
> Ricardo
>
> Have you tested g729 betwe
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
an try to apply Michael's patch... for me 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-User
codec works with all Digium
cards."
Can somebody tell me please?
Thanks,
Ricardo Villa
http://www.telesip.net
there something that can be fixed on the CVS
version to prevent this problem?
Thanks,
Ricardo VIlla
http://www.telesip.net
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