Hi!
> A better option and one Asterisk desperately needs is some kind of
> --lint option,
> Which would check the config for errors and useless misspelled options.
>
>
> I personal find one or more typos or misspelling a month, On my PBXs.
Yes, indeed, same for me. My advice is to always do an
A better option and one Asterisk desperately needs is some kind of
--lint option,
Which would check the config for errors and useless misspelled options.
I personal find one or more typos or misspelling a month, On my PBXs.
Eric Wieling wrote:
Maybe someone will write a patch to print an error
On Saturday 17 January 2004 15:44, Olle E. Johansson wrote:
> Dustin Goodwin wrote:
> > I did find something interesting. If you set reinvite=yes then *
> > can setup the RTP stream so that it avoids the media proxy in the *
> > box completely. I haven't tested to see if it changes anything.
>
> Ca
> Can we please kill "reinvite" - it does not exist in the SIP channel as an
> option for anything. Period.
>
> There is an option called "canreinvite" that you can set to yes or no.
> Setting "reinvite" to anything will not change anything at all.
Olle,
I thought I was the only that was loosin
Maybe someone will write a patch to print an error to the console if
reinvite= is found in the config file.?
On Sat, 2004-01-17 at 15:44, Olle E. Johansson wrote:
> Dustin Goodwin wrote:
>
> > I did find something interesting. If you set reinvite=yes then * can
> > setup the RTP stream so that i
Dustin Goodwin wrote:
I did find something interesting. If you set reinvite=yes then * can
setup the RTP stream so that it avoids the media proxy in the * box
completely. I haven't tested to see if it changes anything.
Can we please kill "reinvite" - it does not exist in the SIP channel as an
o
I did find something interesting. If you set reinvite=yes then * can
setup the RTP stream so that it avoids the media proxy in the * box
completely. I haven't tested to see if it changes anything.
- Dustin -
[EMAIL PROTECTED] wrote:
I am experiencing a problem that from list archive it appears
> I am experiencing a problem that from list archive it appears others are
>
> running into. When I dial from Cisco 7960 via the * to Free World
> Dialup
> destinations that supports G.729 the call fails. The major error from
> the debug log is
>
> Jan 15 00:11:14 NOTICE[22545]: channel.c:1481
I am experiencing a problem that from list archive it appears others are
running into. When I dial from Cisco 7960 via the * to Free World Dialup
destinations that supports G.729 the call fails. The major error from
the debug log is
Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_for
