report any findings.
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If you want to listen into the Developers Summit meeting at Astricon you
can call 972-961-7666 or IAX2/[EMAIL PROTECTED]/4569.
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functions.
http://www.oreilly.com/catalog/asterisk/
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it's only talk.
Also, Digium was just given a few truck loads of cash for BEING such an
open-source driven company.
Stop trying to bad mouth Digium with your so-called rumors.
Jeremy McNamara
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his, or if it's only talk.
Disclose who you heard this rumor from - Then inform them that Asterisk
source has been released into the GPL. If you don't understand what this
means, hire a lawyer to explain it to you.
Jeremy McNamara
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e at every
possible second? If you really need this information, submit a manager
event, then have an another process queue and insert the data into a
database.
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to the channel H.323 and
to Asterisk? (the function OnClosedLogicalChannel in ast_h323.cpp
doesn’t seem to notify anyone!! ).
I cannot answer those questions, but you should be able to tell Open
H.323 to deal with those situations.
Jeremy McNamara
28th.
Then why does documentation for CCM 3.1 show it?
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_data_sheet09186a0080091d7c.html
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Sergio Chersovani wrote:
Jeremy McNamara ha scritto:
Those are SCCP defined constants, which are also defined in ethereal,
which is prior art to chan_sccp.
and now a little warning.
The skinny protocol does not handle the busy status line lamp (hint).
I did it in the chan_sccp code with
Sergio Chersovani wrote:
Jeremy McNamara ha scritto:
Those are SCCP defined constants, which are also defined in ethereal,
which is prior art to chan_sccp.
Pay attention Jeremy, I can analyze here the patch line per line and I
can prove (as I did before) that the whole patch is based on
august 2005)
Those are SCCP defined constants, which are also defined in ethereal,
which is prior art to chan_sccp.
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DISCLAIMED chan_h323 and chan_skinny as well as
countless hours of debugging, testing and performance analysis of
Asterisk. Along with real-world, production deployment of thousands, if
not more, solutions based on Asterisk.
Jeremy McNamara
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beyond a reasonable doubt.
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of trying to work together, all you did was make accusations and
then refused to back it up until I pushed the issue.
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Sergio Chersovani wrote:
ahah, well you have to add "based on the chan_sccp button template
function" on headers of chan_skinny.c
So why in hell did you not offer this how many centuries ago?
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future, but at this time other projects have my attention.
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Ben Klang wrote:
To find out more about this project or to download the new release
please visit the pages at http://projects.alkaloid.net.
Good work, but hopefully you support more than the obsolete v1.0 branch,
as your website documents.
Jeremy McNamara
Steve Totaro wrote:
I believe the NEC IPK VoIP speaks MGCP and NEC has a HUGE portion of the
PBX marketshare.
The last I knew Asterisk's MGCP was the 'old school' standard, which is
not going to cut it for enterprise level deployment of an MGCP solution.
Tristan Graham - Skymarket Ltd wrote:
Is there a better way to go about doing this or should I make up a
suggested patch for people to take a look at ?
Use a MeetMe.
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Innocent Evil wrote:
Hi,
I have complied asterisk from source.
I am trying to recompile app_dial.c after made couple of little changes.
What I want to do is to recreate app_dial.so without recompile full asterisk.
How can I do it?
With a proper Makefile.
Jeremy McNamara
it in a production
environment because it is not distributed with Asterisk.
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Olle E. Johansson wrote:
We can do everything, but the real question is how many database actions
we want per call/registration?
ZERO.
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x27;t port
to realtime in a second
That's what you get for creating a broken implementation - Realtime is
not the answer either.
Be smarter than what you are working on, for a change.
Jeremy McNamara
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As
Roy Sigurd Karlsbakk wrote:
unfortunately, this didn't fix the bug
So do you enjoy being about a year behind in Asterisk development?
Run Asterisk v1.2Beta1.
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Matt Fredrickson wrote:
Good report, lots of information. See if you can reproduce it in CVS-HEAD
(Asterisk, libpri, zaptel)
Yes - After about 1500 to 2000 calls.
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ps working permanently, or just till you discover the
'ifconfig eth0 hw ether xx:xx:xx:xx:xx:xx' command?
Or you very simply call Digium to re-issue the license.
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t the interval to a very
long interval e.g. 1
; or 'never' to disable *entirely*.
;
;resetinterval = 3600
Also, this is a users question, direct future questions to the
appropriate forum.
Jeremy McNamara
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Brian Capouch wrote:
Lots of these same errors, references to "ast_copy_string" but it
can't be found.
Have you done a make clean ?
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Steve Murphy wrote:
Just now, I CVS updated, built, and am running asterisk, and
when we pick up phones and dial, the asterisk console
reports:
Start asterisk with -vvvgc and examine for messages.
Jeremy McNamara
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rtp.cxx, line 416, Error=22
bort, ore dump, gnore? Jun 19 11:41:51 WARNING[29174]:
chan_oh323.c:4202 oh323_release: Forcing the release of entry 0 (call
'ip$localhost/15450').
Jeremy didn't write that channel driver.
Find a clue. Then ask informed questions.
Mojo with Horan & Company, LLC wrote:
I have been using the chan_sccp driver for two months now, and am
highly impressed with the workability and functionality found therein,
just fwiw.
If the author(s) would disclaim their code, it could be included with
Asterisk.
Jeremy McNa
Zoa wrote:
I used the chan_sccp lately and it seems to work correctly with the 7920
This thread is not about chan_sccp.
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Andreas Czerniak wrote:
Dear,
this problem was solved by an patch by Jose Carlos Garcia Sogo:
http://www.beronet.com/bugs/bug_view_page.php?bug_id=046
And it only occurs in asterisk stable versions not in the head revision.
*Gasps* "Stable" code crashing
Jerem
Ganbold Tsagaankhuu wrote:
It seems like asterisk can't find client.
How can I make asterisk accept h323 call to sip phone?
Configure Asterisk properly.
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Senad J wrote:
Just compare it to SIP, and you will see HUGE differences :)
Because SIP is the new H.323.
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To
Eric Wieling aka ManxPower wrote:
Will this code be disclaimed so it can be added to CVS-HEAD?
Digium BOUGHT it. I would hope it would get added to cvs -head.
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something to do with it. This channel driver from ObjSys compiles,
loads and actually makes a call. Which is more than I could ever get
your driver to do on the same box.
So instead of filing a bug report you simply bitch about it?
Go back to your hole.
Jeremy McNamara
Brian West wrote:
Autoconf and pals have been offered up twice totally working and they
were going to help maintain it.. but in the usual fashion Digium and
gang ran that guy off.
Help or guarantee to be dedicated to supporting the crap for the next
100 years?
Jeremy McNamara
This needs to be posted on http://bugs.digium.com and you need to submit
a Disclaimer to Digium before I can commit it.
Thank you,
Jeremy McNamara
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Jerry Hicks wrote:
I know it's a real PITA but has anyone considered autoconf, automake,
et al?
Are you volunteering to support it for the next 100 years?
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your cross compiler instead of just whatever gets
found in your PATH.
I have pondered tweaking the various makefiles to make this process
easier, but I have not gotten that motivated, yet.
Jeremy McNamara
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or testing and review.
We have this code already setup on a test system with PRIs to the PSTN.
If you happen to be one of our customers and want access, please contact
either Greg or myself so we can enable your account on this test system.
Jerem
Michael Giagnocavo wrote:
Sure, realtime is one way to do this. Doesn't mean it's the only way that
should be considered.
Exactly. Realtime is not the answer.
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n the future
(then again, I might just spend a couple of hours and get my own test
server fixed so it can run valgrind ).
Yes, we have all of our test gear accessible via public IP addys.
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re to be a module? Then anyone can create their own method of
communication as they see fit.
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Daniel Pocock wrote:
Try the open source implementation and let me know if you have the
same problem:
There is no OPEN SOURCE implementation of G.729.Refrain from posting
illegal links.
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Diseyi Diffa wrote:
however i still get the same error as above. Does anyone no how i
can fix this problem. Any help will be greatly appreciated.
Run ldconfig.
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with the way you linked the C++ code.
Why oh why couldn't you have written this in C, like a sane person would
have done.
Jeremy McNamara
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Tom Ivar Helbekkmo wrote:
Just had a core dump...
#0 0x080596cd in ast_queue_frame (chan=0x828a400, fin=0x9d38)
at channel.c:412
Have you done a make clean lately?
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are wrong? If did have even a
shred of evidence you would have been spewing it all over the place.
Leave us alone, I'm sure i'm not the only person that is tired of
deleting your email.
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Sylvain Munaut wrote:
And here's the patch for indications.conf.sample
Good work. Now submit this this to http://bugs.digium.com
Thank you,
Jeremy McNamara
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the current cvs -head (if my memory
services me)
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[EMAIL PROTECTED] wrote:
You need to go to logger.conf and send the "debug" level somewhere.
and issue a logger reload command in the Asterisk CLI.
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Richard Neese wrote:
the answer is rtc and zaprtc should be put right into *
we need real time timing...
Zaprtc does not play nice with SMP based systems.
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compiled and in place for hardware devices I don't see myself
using.
Simply call Digium (877-Linux-Me) for an X100P, after all they GAVE US
Asterisk for nothing, then noload what you don't want in modules.conf.
Problem solved.
Jerem
ariant.
Say what?
With todays CPU power there is absolutely no need for expensive DSPs.
This is why Digium didn't design hardware with DSPs on board.
Jeremy McNamara
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tatic.
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ns do not scale. Trying to process CSV's are completely
wrong and triggers are going to be a loosing battle for CPU time as your
volume increases.
AGI and/or RADIUS are not the solution either. Use the power of
Asterisk, don't side step it.
Olle E. Johansson wrote:
Jeremy, is H.323 able to use SRTP?
chan_h323 uses asterisk's RTP stack, so if it supports SRTP, chan_h323 will.
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Federico Alves wrote:
Do we have any idea when the H323 rewrite will be available on the CVS?
When its ready to be commited to cvs.
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brian k. west wrote:
Why aren't the queries a config option?
Or at least break out a configurable table name
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James Courtier-Dutton wrote:
Jeremy McNamara wrote:
There is no "proprietary" version of Asterisk. If you want G.729
you have to pay Digium (and Digium pays the patent holders) for the
non-gpl'd code that codec_g729b.so utilizes. Also, if Asterisk's
copyright is not c
Cristian Manoni wrote:
Hi all
i have an asterisk crash every time the number of channel CAPI arrives at
490
CAPI[contr1/541774629]/490.
Feb 2 11:36:37 WARNING[10251]: Unable to allocate socket: Too many open
files
Feb 2 11:36:37 WARNING[10251]: Unable to create RTP session: Too many open
files
F
Linus Surguy wrote:
If it was your intent, please say so, and we can make a note to manually
patch it our end each time we checkout as we need the callerid= bit to
always work.
If you want to override callerid use SetCallerID on the appropriate
exten line(s).
Jeremy McNamara
.voip-info.org/wiki-Asterisk+E164+Call+Routing
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What does this patch do that nat=yes doesn't accomplish?
Also, this should be submitted to http://bugs.digium.com/ not the
mailing list.
Jeremy McNamara
William Waites wrote:
This is not thoroughly tested, but seems to work for me. It consists in
three changes:
1. introducti
READ THE README
Jeremy McNamara
Garry Adkins wrote:
Hi,
I'm on a RH 9 box, fresh install.
When I try to compile channel h323, I get multiple compile errors. Can
someone help?
asterisk, ptlib, openh323 all are fresh from CVS.
Thanks!
Here's what I've done so far:
All the so
Quicknet hardware is broken, don't bother. Support Asterisk, buy Digium
hardware.
Jeremy McNamara
[EMAIL PROTECTED] wrote:
Hi
I have guessed from (not having any replies on) asterisk-users, that there
is no way to use hardware codec like G.723 on LineJack. I was thinking it
should be p
John Todd wrote:
Since we seem to be moving the power of the phone system further out
into the hands of the users, it does make sense that a Web GUI would
allow them to control their own telephony configurations.
How to you plan to deal with the various stupid things users are going
to do?
Add this as either a bug or feature request to http://bugs.digium.com.
Jeremy McNamara
[EMAIL PROTECTED] wrote:
weird Ive been trying to send this mail for hte last few weeks and it
isnt getting through... another repost..
According to this document,
http://www.ietf.org/internet-drafts/draft
gium and at this point my operation
is not far enough along to support both termination customers and
customers running our billing system.
Jeremy McNamara
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I seriously hope you didn't waste your time implementing a RADIUS hack
into Asterisk. It is absolutely, positively not necessary.
Jeremy McNamara
Lubomir Christov wrote:
Hello,
we are ready with a Radius channel for asterisk. It can do Radius
authentication and accounting. I think
n't match
with them, yet
Jeremy McNamara
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Luke Howard wrote:
Might want to get rid of the linefeeds...
They should be gone now. Right?
Jeremy
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[EMAIL PROTECTED] wrote:
are we going to keep conf.sample in the channels/h323 directory, or should
this be moved into the asterisk/configs
It prolly needs to be moved.
Be advised this will NOT compile with the rest of Asterisk, at this
time. If your really motivated to make it work, before
Of the top of my head:
H.323
---
OpenPhone
OhPhone
GnomeMeeting
SIP
---
M$N ( < v5 )
sjphone
Jeremy McNamara
Steve K wrote:
Other than Gnophone, what are some names of other
clients that can be used with an Asterisk server?
Steve K
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