A number of people are reporting that Safari is not working properly with JIRA.
Use Firefox or Chrome for now.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806 - USA
www.digium.com -=- www.asterisk.org
in jira ?
No. We migrated as much as we could. This was one minor thing that was
not migrated over.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806 - USA
www.digium.com -=- www.asterisk.org
On 06/02/2011 06:46 AM, Terry Brummell wrote:
We use Jira at work. I hate it. Hope you have a better experience than
I've had!
We've been using it for years internally to Digium. We've been happy
with it.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445
rather report
something via email, email espiceland at digium dot com and me.
Thanks,
[1] http://lists.digium.com/pipermail/asterisk-dev/2011-May/049088.html
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806 - USA
and if so I'll raise the issue...
Try the latest code from the 1.8 branch. This sounds very familiar. I think
it has already been fixed.
$ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8 asterisk-1.8
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445
view it as if there is a problem in
the code, it is _much_ less expensive to get it resolved in up front peer
review as much as possible than later on once users encounter a bug, report it,
developers debug, fix, and test. That's the tradeoff.
--
Russell Bryant
Digium, Inc. | Engineering
it is
now my full time job, I still put many personal hours into the project. I care
very deeply about the success of Asterisk. I truly believe that the steps we
have taken with release management are in the best interest of the project.
Thanks,
--
Russell Bryant
Digium, Inc
- Original Message -
On Apr 28, 2011, at 9:53 AM, Russell Bryant wrote:
I don't think it's a separate issue at all. I would like to see
discussion of exactly which issues are preventing users from using
Asterisk 1.8. We're trying to shift focus to those issues and get
them
- Original Message -
On Apr 28, 2011, at 10:21 AM, Russell Bryant wrote:
For us the biggest issue is multi-tenant parking not working. We've
really given up testing anything beyond that point because without
that feature there really isn't any way we could use it.
Broken
the
security maintenance period. It has to be pretty clear, though, and in this
particular case, it is.
Can you point to the bug number please? I want to make sure this voicemail
problem is resolved as soon as possible.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
uptodate review https://reviewboard.asterisk.org/r/1185/
Thanks, Alec. I have added this to the roadmap for the next 1.8 update. I'll
make sure it gets resolved before then.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL
priorities. Officially
letting off some of the pressure from older branches does help. I would like
to be making faster progress through bug reports and patches. I do have an
open position for another full time Asterisk developer at Digium in case anyone
is interested. :-)
--
Russell Bryant
to version compatibility can be found in the
UPGRADE*.txt files in the Asterisk source.
http://svn.asterisk.org/view/asterisk/trunk/UPGRADE-1.8.txt?view=markup
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806
and Packages project
on the Asterisk issue tracker, http://issues.asterisk.org/.
Thanks,
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Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806 - USA
www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org
tracking system and forgot to
disable email first.
Thanks,
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Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW -Huntsville, AL 35806 - USA
jabber: rbry...@digium.com-=-skype: russell-bryant
www.digium.com -=- www.asterisk.org
the problem.
Thanks!
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Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW -Huntsville, AL 35806 - USA
jabber: rbry...@digium.com-=-skype: russell-bryant
www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org
watch' the console output.
It's asterisk -r or asterisk -rx. The message that says remote unix
connection means a remote connection to Asterisk over the UNIX domain socket,
which is what the remote console uses. AMI connections have a different
message associated with them.
--
Russell Bryant
, as well as what distro you are using, and the
version of gcc that you have (gcc --version).
Thanks again for the feedback,
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW -Huntsville, AL 35806 - USA
jabber: rbry...@digium.com-=-skype
. There is
also the ability to enable DTMF key presses to swap between spy, whisper, and
barge modes.
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Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW -Huntsville, AL 35806 - USA
jabber: rbry...@digium.com-=-skype: russell-bryant
www.digium.com
context for that parking
lot. For example, if you set up the parking lot in features.conf as:
[parkinglot1]
context = parkedcalls_custom
parkpos=800-850
the hint should be:
exten = 800,hint,park:8...@parkedcalls_custom
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
that is not supported in any current version of
Asterisk. However, a large amount of work has gone into connected party
ID support which will be included in Asterisk 1.8. I expect the first
beta of 1.8 to be available this month.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan
on this module
again. It was something I had done in my free time. I got it working for me
and never got back to it to address problems other people have reported.
Hopefully another developer will take interest and try to help.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
/pipermail/asterisk-dev/2009-December/041336.html
[4] http://www.asterisk.org/asterisk-versions
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Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
can tell from looking over the source code :-)
Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com
this information on http://bugs.digium.com/ and we'll help
you resolve the issue.
Thanks,
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Russell Bryant
Digium, Inc. | Senior Software Engineer, Open Source Team Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
the timers.
In 1.6.1, this should not be required. It's probalby a check in the
code that shouldn't be there anymore. If you post this on
bugs.digium.com, I'll remove it.
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Russell Bryant
Digium, Inc. | Senior Software Engineer, Open Source Team Lead
445 Jan Davis Drive NW - Huntsville, AL
Greetings,
We recently moved our public subversion mirror to a new server. It is
currently down for maintenance while we resolve some unforeseen
problems. It should be back up by the end of the day.
I apologize for the inconvenience,
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Russell Bryant
Senior Software Engineer
Open Source
, thread-iosin, sizeof(sin));
if (res sizeof(*mh)) {
- ast_log(LOG_WARNING, midget packet received (%d of %zd
min)\n,
res, sizeof(*mh));
return 1;
}
if ((vh-zeros == 0) (ntohs(vh-callno) 0x8000)) {
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Russell Bryant
Senior Software Engineer
Open
the subscribe_event, asterisk starts as normal.
What version of openais are you using? The versions listed in the
stable section of openais.org do not include a bug fix to the event
service that prevent a crash. Try one of the newer versions listed on
the site.
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Russell Bryant
Senior Software
some development effort dedicated to
getting completed.
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a backtrace from Asterisk when it is
hanging. If it is just failing to start, I would need to see the full
Asterisk console output on startup to see what happens when it decides
to stop.
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc
(system:capture_1))
[macro-connect]
exten = s,1,Wait(3)
exten = s,n,System(ast_connect)
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? That may help me
understand the audio path involved ..
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earlier, more info here ...
http://www.russellbryant.net/blog/2008/01/13/jack-interfaces-for-asterisk/
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that's useful, and it's also more
difficult to implement with the current code.
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AstriCon 2008
/01/13/jack-interfaces-for-asterisk/
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Register
a video?
This is not something that is supported right now. However, it would
be relatively straight forward to add for a developer interested in
adding it.
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc
On Sep 8, 2008, at 7:31 PM, Russell Bryant wrote:
On Sep 8, 2008, at 9:15 AM, Gordon Henderson wrote:
Does/Will asterisk support video streaming on hold?
Been playing with videphones as of late, and a client asked about
video on
hold - standard MoH works fine - but on the target video
, and an
optional arg, that will send a tone to the second channel.
see main/features.c
Thats good to know.
Will the xml-over http manager interface be able to do it too? (pretty
please?)
Yes.
Also, in addition to being a manager action, it is also a dialplan
application.
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Russell Bryant
Senior
fine with any phone that supports that codec.
Personally, I have only used it with Polycom phones. Also, again,
Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has
full support.
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Russell Bryant
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Digium, Inc
version. So, please give 1.4.22 a try. Then, please gather details
and post them to http://bugs.digium.com/. If you'd like to discuss
what you need to do to create the bug report, join #asterisk-bugs on
the freenode IRC network.
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
questions about it and Asterisk, I would be happy to try to answer them.
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Russell Bryant
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Digium, Inc.
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things have been
fixed since 1.6.0-beta9.
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Russell Bryant
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Digium, Inc.
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any suggestions that would make app_jack or chan_console
easier for you to use, then please let me know. Feel free to contact me
directly.
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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. This assistance is free when you are using Digium hardware.
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Digium, Inc.
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use the menu to see if the dependencies have been met, or to
disable it from being built and installed.
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subscribe but still expect MWI. Some
phones may freak out if they receive MWI when they haven't subscribed to it.
Thanks again for clarifying! I appreciate it!
You are quite welcome.
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc
Philipp Kempgen wrote:
Is subscribemwi valid in peer context only or also in general?
sip.conf.sample is not clear about that.
Only within a peer definition.
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc
Asterisk to _only_ send MWI with an associated subscription.
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Russell Bryant
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still expects MWI to be handled in an event based fashion.
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Open Source Team Lead
Digium, Inc.
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of voicemail. There
would not be any easy way to trigger the poll after someone has made
changes using their IMAP client.
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). You can grab it directly
from svn, as well.
$ svn co http://svn.digium.com/svn/asterisk/branches/1.4
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Russell Bryant
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to access it as
opposed to the astdb functions.
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Russell Bryant
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everything works great, just like its supposed
to with imap.
Any ideas?
It looks like app_voicemail is failing to load when you build c-client
with SSL support. Try running module load app_voicemail.so from the
Asterisk CLI to see what the error message is.
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Russell Bryant
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://svncommunity.digium.com/svn/russell/asterisk-1.4/
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Russell Bryant
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section of sip.conf, set allowguest = yes. Also, set
context = something, where something is a context that you would like
unauthenticated calls to have access to.
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc
memory show summary and memory show
allocations, which show you all of the memory allocations that Asterisk
has made by file, function, and line number.
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Russell Bryant
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Digium, Inc.
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1.6.0 branch. Also, make
sure you're using a reasonably current version of OpenSSL.
$ svn co http://svn.digium.com/svn/asterisk/branches/1.6.0
If you still have trouble, feel free to report it on
http://bugs.digium.com/.
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium
.
However, when your friend calls you, and Asterisk makes a call out to
your client, it offers encryption, and your client accepts it.
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Russell Bryant
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when the DTMF follows the media path.
Nobody can help any further without seeing the details of your
configuration.
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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, you would have the Set()
command in the dialplan (extensions.conf).
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Russell Bryant
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Digium, Inc.
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AstriCon 2008 - September
.
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Russell Bryant
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Digium, Inc.
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Register Now: http://www.astricon.net
asterisk
On Aug 8, 2008, at 4:48 AM, Stefan Gofferje wrote:
Hi,
addons 1.6 don't compile here. Any ideas?
It looks like you're trying to compiled Asterisk-addons 1.6 against
Asterisk 1.4. You will need to install Asterisk 1.6 before you can
compile and install Asterisk-addons 1.6.
--
Russell
On Aug 8, 2008, at 7:39 AM, JR Richardson wrote:
Asterisk 1.6 currently has T.38 origination and termination support.
It does not yet have fax gateway support.
--
Russell Bryant
Russell, Can you please clarify what you mean. I think there is
still a bit
of confusion as to what
-1.6.0$ ./configure --with-asterisk=/path/to/asterisk-1.6.0
However, as Tzafrir noted in another reply, it is worth mentioning that
regardless of which method you use, Asterisk-addons 1.6.0 modules _must_
be used with Asterisk 1.6.0.
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Russell Bryant
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.
You have yet to bring any useful discussion to the table.
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channel _absolutely does NOT_ remove the need for
locking to synchronize access to channel data structures.
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be an extremely useful new
feature to have, but as fair as I know, is not currently available.
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Digium, Inc.
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asterisk-users
with
the latest 1.4 versions of Asterisk. A _lot_ of work has gone into IAX2
support in Asterisk 1.4, and specifically, the most recent 25% of the
1.4 series or so.
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this works? im using 2 xlite,
Just a hunch ... if you're using xltite, it's likely that you're not
pressing the digits fast enough to satisfy the default timeout. The
default featuredigittimeout is 500 ms. Change this option in
features.conf and increase it to 2000 ms and try again.
--
Russell
Steve Davies wrote:
Does this mean that the fixed IAX security fix for 1.2.28 (1.2.28.1?)
will also be officially released now?
If it helps, I have given 1.2 trunk some light testing and it seems
reasonably sane.
Thanks for the reminder. I will build that release right now.
--
Russell
before there's a response :)
Hrm...looks like I was right so far :)
That is correct.
You can also put some of those options in /etc/asterisk.makeopts or
~/.asterisk.makeopts if you want to set system-wide or per-user options for all
Asterisk builds done on the system.
--
Russell Bryant
, but it looks like quite a big maze.
Channel names are not guaranteed to be unique at all. However, all channels
have a uniqueid associated with them. You can access it in the dialplan via
${UNIQUEID}.
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc
hangup a given channel?
While channel names are not unique over time, at any given point in time, there
should not be more than 1 channel with the same name.
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Digium, Inc.
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using? I have made a lot of G.722 related
fixes over the last few months.
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device as what app_queue
should look at before attempting to call the agent.
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seen any code, or demonstrations showing that the code may exist. So, it
is not supported in any version of Asterisk, and I have absolutely no idea when
and if it ever will be.
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://www.asterisk.org/node/48360
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can _not_ simply concatenate two [wav]
files. :)
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that describes what you have
found. We will help you get the fix into svn.
Thanks,
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that, I just use a nice tool like audacity
...
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://svncommunity.digium.com/svn/russell/asterisk-1.4/
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should see it, compile it, and
install it as usual.
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the source code/files.
I want to rebuild them for my system using Callie's voice.
All of the sounds releases include a text file with the script for every sound
file.
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Digium, Inc
that to hopefully notify everyone
that doesn't closely monitor commits to Asterisk, or other high volume Asterisk
mailing lists.
Thanks for the feedback,
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in both servers, and in this situation if I make a call
from B to A, suddenly peers in server A are able to call peers in machine B.
Try using the DUNDi query CLI command to see what results your server is getting
when you try to make calls.
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want
pbx_config, chan_sip, app_dictate, app_dial, probably some others ...
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for an established T.38 passthrough call.
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will) increase latency for call completion, will
increase bandwidth consumption, among other things.
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asterisk
of the code since 1.4.15. I would
suggest trying the latest version. If it still gives you trouble, please let
us
know on http://bugs.digium.com so that we can fix it up for you.
Thanks,
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc
Anthony Messina wrote:
Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing
out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3.
I get the following:
This should be fixed in Zaptel 1.4.9.2.
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Open
=markup
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'
The arguments to System() are a bit different. Put it in just like you would
type at the command line.
System(/tmp/netcid.py 2000 Joe)
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exactly what
they are. You can always compare them with diff.
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to, is it another SIP phone? If so,
what is the associated configuration in sip.conf? Do you have call-limit set
to
some value, or the combination of callcounter and busylevel? If so, what are
they set to? (You must have these options set for it to work)
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[EMAIL PROTECTED] wrote:
Am I doing something wrong? What I should do to get ooh323.conf
cp asterisk-ooh323c/h323.conf.sample /etc/asterisk/ooh323.conf
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Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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and both asterisk and mysql are located in the same
server. What do the message mean? It seems the message will cause
the user failed to login. How can it be solved?
Did you install res_config_mysql from asterisk-addons?
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc
]
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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asterisk-users mailing list
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http://lists.digium.com/mailman
.
Yes, that message generally indicates a deadlock in Asterisk.
Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846)
There have been 380 fixes made to Asterisk 1.2 since 1.2.14, 61 of which were
made to chan_sip. :)
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Russell Bryant
Senior Software Engineer
Open Source Team
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