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Someone needs to change the HPEC installation instructions.
Took me a while last night to figure out what was going on :)
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Matt Riddell
Director
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be able to
respond as he wrote the patch.
Are we fine using AGI in this situation? If so, I'll update the wiki to
be a bit more verbose.
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Matt Riddell
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system.
Regardless without the #if...#endif I was able to compile without
problems.
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to ask
where 1.3 went to?
1.3 was trunk once 1.2 was released. 1.5 is now trunk.
As far as I understand.
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.
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Matt Riddell
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well,
and that they will therefore be creating (or are in the process of
creating) their own GUI.
I will post information as I find it.
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James Jones cross posted.
DO NOT CROSS POST. THIS HAS ALREADY BEEN ANSWERED IN ASTERISK-USERS!!!
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be
something in this, or if it's only talk.
I too would like to know where you heard this or whether you just made
it up...
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is played)
Author: bweschke
Date: Sun Sep 3 15:44:14 2006
New Revision: 41916
URL: http://svn.digium.com/view/asterisk?rev=41916view=rev
Log:
Fix enum indexing problem with m() in WaitExten. Reported by Pavel J,
in asterisk-dev.
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or changes to existing
ones, even if just to see where the activity is and to pass on requests
for testing to the community.
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Matt Riddell (IT) wrote:
This is a little off topic, but does anyone know if there is a link to
receive an RSS feed of the changes to the bugtracker?
I know there is one for the comments front page, but it would be really
cool to be able to see
screw up
on my part
:)
Any help would be much appreciated!
:)
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Russell Bryant wrote:
On Sun, 2006-08-06 at 17:17 +0200, Matt Riddell (NZ) wrote:
I'm trying to let the manager send jabber messages
usr/src/asterisk/manager.c:889: undefined reference to `ast_aji_get_client'
The problem is that this function
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Dave Cotton wrote:
You have 2 choices:
1) Do the work yourself
2) Pay for someone to do it for you
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No Matt you've got it wrong Decartes didn't say Je pense donc je suis
he said Je râle donc je suis
..
Have a look for app_conference.
Hint: iaxclient.sf.net
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this apply to all realtime cdr
operations i.e. postgres, odbc etc or just MySQL?
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,
Matt Riddell
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for the patches?
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- really a *-users question though.
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?
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Jerris, Michael MI wrote:
Matt Riddell
harry gaillac wrote:
Hello,
Is it possible to provide chan_exosip2 for both asterisk
and openpbx
?
Only if the authors are happy to disclaim the code to Digium.
Disclaiming chan_exosip would make little difference as it is based on
GPL
[EMAIL PROTECTED] wrote:
On Mon, 14 Nov 2005, Matt Riddell wrote:
Tilghman Lesher wrote:
On Thursday 10 November 2005 03:38, Chih-Wei Huang wrote:
Chih-Wei Huang wrote: BTW, my G.723.1 and G.729 codec are from Intel
IPP library.
I decided to file a report to Mantis:
http://bugs.digium.com
shenanigans wrote:
I was interested in getting feedback from current mail group users.
This is really pushing it now. That's three times.
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Martin Mateev wrote:
it's friday night, at least here, get a life
Um...it's Sunday here (5:13am) and I'm still working.
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Kevin P. Fleming wrote:
Matt Riddell wrote:
Sorry, where is this Kevin?
Look for VM_CATEGORY in the source, it's a channel variable.
I don't see anything related to it in voicemail.conf.sample as of 30
seconds
ago :)
It's not in there, because it doesn't belong
something custom if enough people were
interested.
Any votes for an Open Source development of a phone with Open Source firmware?
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about dial?
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Michael Giagnocavo wrote:
Yea, I'd recommend people to read Code Complete (now in second addition)
that would back up a lot of what you say with hard data from a lot of real
projects.
I'd second that recommendation even though the book came out of the
Microsoft Press!
:)
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as the 1.0.x track?
roy
/frustrated
What kind of response do you expect to that?
Why not just post the bug id's of the bugs you have found (preferably
with patches).
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Brian West wrote:
I just spoke with mark and we are all gonna get together and come up with
MeetMe2 (it will be in addons for a while)
*PHEW!!*
(The sound of a collective sigh of relief)
:)
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to go down to the studio at the moment, but
I've got a full on editing suite at home. I've been doing the sound
engineering/recording/mastering thing for like 15 years now so I guess I
have some knowledge. At least I'd hope so! :-)
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Anton Tinchev wrote:
Matt Riddell wrote:
Anton Tinchev wrote:
Will be there new card?
I'm asking it, 'couse i'm going to buy 3-4 cards?
Or i should wait for the new one?
I'd buy now. There has so far been no information on the new cards
from Digium (I.E. no confirmation that the card will exist
-users question as you are wanting to
_use_ Asterisk.
1) Turn on qualify (i.e. put qualify=2000 in sip.conf for each entry)
and then you'll be able to see which is causing the delay.
2) Try out an IAX softphone...
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) and the current cards have been proven to work well.
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as opposed to 8 x the_tone_specified x the_length
specified.
Kind regards,
Matt Riddell
CEO http://www.sineapps.com
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Update of /usr/cvsroot/asterisk/channels
In directory localhost.localdomain:/tmp/cvs-serv27553/channels
Modified Files:
chan_vpb.c
Log Message:
/ check so as not to enable loo-drop on FXS
??? A loo drop on FXS? Doesn't sound too nice!
Matt Riddell
from 4K to 16K, this
has helped some people.
Matt Riddell
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