Hi all,
I would like to have the caller in app_queue to hear inband call progress
ringing instead of music on hold. Using options r will
enforce false ringtone which is not what I want, I want the app_dial call
progress forward to app_queue instead. Can anyone give me some
hints on
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with the changes to the ztdummy to rely on rtc vs jiffies, I am now
forced to increase the interrupt frequency time by roughly 10x the
frequency recommended for the SMP processing systems.
Is this wise? Or would it be better to not assume that the
Hello All,
I have a question which probably sounds silly.
I am using Asterisk SVN-branch-1.2-r8140 built. To be honest I am still
confused with this SVN things. The main reason I use it is that I think I can
get the bugs fixed by updating it regularly, which I previously could not do
that
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Sean Cook wrote:
with the changes to the ztdummy to rely on rtc vs jiffies, I am now
forced to increase the interrupt frequency time by roughly 10x the
frequency recommended for the SMP processing systems.
Isn't that backwards? Using the RTC means we are _not_ relying on the
frequency of
Aryanto Rachmad wrote:
I have a question which probably sounds silly.
I am using Asterisk SVN-branch-1.2-r8140 built. To be honest I am still
confused with this SVN things. The main reason I use it is that I think I can
get the bugs fixed by updating it regularly, which I previously could
Don't over estimate my familiarity with the code :-)
I think I see something odd in channel.c, in code not touched
by this patch. In ast_activate_generator there is a call to
ast_settimeout(chan, 160, generator_force, chan);
Now it might be just me, but should the 1st and 4th parameters
be
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or, via email, send a message with subject or body 'help' to
[EMAIL PROTECTED]
please stop sending me these emails..brian bellAt 12:53 PM 1/20/2006,
you wrote:
If
Asterisk 1.2 was released over 1 year after Asterisk 1.0, which resulted
in many users trying to run the development version of Asterisk in a
production capacity so that they could take advantage of the new
features that had been added. This produced a flurry of extraneous bug
reports and caused
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Message: 6
Date: Sat, 21 Jan 2006 02:42:23
What about using trunk -- will that get me 1.2.2 or netsec of both?
on Friday 01/20/2006 Kevin P. Fleming([EMAIL PROTECTED]) wrote
Aryanto Rachmad wrote:
I have a question which probably sounds silly.
I am using Asterisk SVN-branch-1.2-r8140 built. To be honest I am still
Tzafrir Cohen wrote:
The following assertions are based on my own knowldge and are hopefully
close to correct. Feel free to comment:
1. The version of Asterisk and the version of libpri must match.
The API should never change between minor releases. 1.2.X of Asterisk
and 1.2.X of libpri
On 1/20/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
ast guy wrote:
here is the ulimit command output.
[EMAIL PROTECTED] ~]# ulimit
unlimited
And the response to ulimit -n
is what?
--
Cheers,
Matt Riddell
'ulimit -n' returned 1024
- - astguy
I upgraded to the latest SVN trunk. No configuration changes were made.
Now I am receiving the following message:
Jan 20 22:02:06 WARNING[30500]: pbx.c:1556 pbx_extension_helper: No
application 'SIPAddHeader' for extension (default, 8080, 11)
Did I miss something, or is this a bug?
Thanks.
How did you upgrade. Did you do a make and then a make install???
What does 'show version' on the CLI show???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Trevor G. Hammonds
Sent: Saturday, January 21, 2006 1:18 AM
To: Asterisk Developers
Sean wrote:
with the changes to the ztdummy to rely on rtc vs jiffies, I
am now forced to increase the interrupt frequency time by
roughly 10x the frequency recommended for the SMP processing systems.
Is this wise? Or would it be better to not assume that the
CONFIG_HZ == 1000 and base
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