Hi folks,
I spotted a bug in handle_request_info: in an if condition the code assumes
to receive NULL on error, while in fact it receives an empty string. The
attached trivial patch fixes this.
Patch is done against chan_sip.c from r8023.
C'ya,
Marc
--
Marc Haisenko
Comdasys AG
Hi all,
I have a problem with Three-Way-Calling on my ISDN Card. I have two HFC
PCI cards and one TDM400 (2x FXS). I wrote already to the users list,
but nobody could help me. So I would like to solve the problem by
myself.
I get the following message if I try to do three way calling (having
On Fri, 2006-01-13 at 11:15 -0600, Aaron Daniel wrote:
My boss and I have been working on a patch to the voicemail code, and
I'd like to see what everyone thinks of it. I'd like suggestions and
stuff on anything that needs to be changed, as this is the first time
we've patched the code,
On Fri, 2006-01-13 at 11:15 -0600, Aaron Daniel wrote:
My boss and I have been working on a patch to the voicemail code, and
I'd like to see what everyone thinks of it. I'd like suggestions and
stuff on anything that needs to be changed, as this is the first time
we've patched the code,
On 1/13/06, Marc Haisenko [EMAIL PROTECTED] wrote:
Hi folks,
I spotted a bug in handle_request_info: in an if condition the code assumes
to receive NULL on error, while in fact it receives an empty string. The
attached trivial patch fixes this.
Patch is done against chan_sip.c from r8023.
Well, I think we're calling it normalize cause that name sorta stuck.
We are in fact just changing the volume of it, and using sox, cause as
you say, it supports more formats. Normalize just didn't do what we
wanted it to, and didn't support the format we were using, so sox seemed
like a much
On Friday 13 January 2006 18:29, BJ Weschke wrote:
Patched. Thank you! In the future, please also check out
http://bugs.digium.com/ for bug reports and patch posting so we've got
a better cyber-papertrail of these types of reports.
ACK.
C'ya,
Marc
--
Marc Haisenko
Comdasys AG
On Fri, 2006-01-13 at 11:45 -0600, Aaron Daniel wrote:
Well, I think we're calling it normalize cause that name sorta stuck.
We are in fact just changing the volume of it, and using sox, cause as
you say, it supports more formats. Normalize just didn't do what we
wanted it to, and didn't
On Jan 13, 2006, at 4:45 AM, Tzafrir Cohen wrote:
Is there any reason, thus, that the version of asterisk-sounds is
automatically bumped on each release of Asterisk? Could it be bumped
only when there is actually a releasble change? If not: could the
tarball include some sort of changelog?
anyone got pointers as to how to write code for sip
or iax2 video feature to a sip or iax2 client software?
Thanks!
Matt
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would you want the patch against trunk or 1.2.1? I wrote it against 1.2.1
but just svn co trunk and massaged it against that tree also. Is there
any particular coding standard and/or etc that is prefered and if so, I
will clean the patch up to make less work on someone else. I'm getting
the
On Fri, 13 Jan 2006 12:28:14 -0600, Steven Critchfield wrote:
Since you are just changing the volume level, is there a reason you
aren't using the record_gain option that I see in the code I just
checked out from trunk? It would save you from spawning a process out of
asterisk for the
On Fri, 13 Jan 2006 11:28:31 -0600, Steven Critchfield wrote:
You aren't normalizing the audio, you are just adjusting the volume. If
you are going to go to the work of using an external app, why not use
normalize so it truly is normalized and doesn't introduce problems when
audio comes from
On Fri, 2006-01-13 at 15:03 -0600, Robert A. Thompson wrote:
would you want the patch against trunk or 1.2.1? I wrote it against 1.2.1
but just svn co trunk and massaged it against that tree also. Is there
any particular coding standard and/or etc that is prefered and if so, I
will clean the
+ status = pbx_builtin_getvar_helper(p-chan,
CHANLOCALSTATUS);
+ if (autologoffunavail status !strcasecmp(status,
CHANUNAVAIL)) {
+ char agent[AST_MAX_AGENT] = ;
+
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