Which version of the driver do you use?
Fernando Herrera wrote:
Hello,
I have installed oh323 channel driver. Outgoing calls to H.323 world do
not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that
userInputMode=RFC2833 has already been set.
Does anyone know how to make
Hello all,
Updated versions of asterisk-oh323 are now available both for use with
Asterisk v1-0 (version 0.6.7) and Asterisk HEAD/v1-2 (version 0.7.3).
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
Hi Tony,
The new packages of asterisk-oh323 (for STABLE HEAD) are ready to be
released on inAccess Networks site. Expect them in the following
two or three days.
Michael.
Tony Mountifield wrote:
I have successfully been using OH323 v0.6.5 with Asterisk 1.0.x.
I now need to move to CVS HEAD
Hi Steve,
Your [general] section looks fine.
In the [register] section remove everything else and leave these lines.
context=incoming-h323-calls
alias=HMA0200.10szxn-
alias=22xx2912
alias=HMA0200.10szxn-
alias=22xx2913
Now all H.323 calls will enter in 'incoming-h323-call' context.
Hello all,
A new bug-fix release of asterisk-oh323 for the *stable version*
of Asterisk is available. This version has the option to compile
with latest OpenH323/Pwlib libraries but we recommend to stay
with the Janus version.
The updated version that is compatible with the *CVS HEAD version*
Mirko Marghitola wrote:
Asterisk don't send the 180 Ringing SIP message to the calling phone
when the called party is ringing. How can I force asterisk to send the
ringing messages? The option 'r' in the Dial() command or the Ringing()
command didn't solve the problem.
Mirko
Did the sip
Papadopoulos Georgios wrote:
Hello,
I am pretty new with Asterisk and I am using it as an H323 gateway.I
would like to keep the same h323-conf-id in the outgoing leg as in the
incoming leg.
So far I have only tried inaccessnetworks' oh323 module, but I think
this is a more generic issue. My
There is nothing wrong with your config, it is just unimplemented
functionality.
Michael.
Alexander Topolanek wrote:
Hi,
Perhaps there's something wrong in my config...
I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting
up an h323 trunk. When dialling into asterisk I
Tony Mountifield wrote:
In article [EMAIL PROTECTED], I wrote:
In article [EMAIL PROTECTED],
Michael Manousos [EMAIL PROTECTED] wrote:
Can you get an ethereal trace on a call with that problem?
Run an RTP analysis on the captured stream (Tools Menu) and save
the contents of the RTP packets
Vamsi Pottangi wrote:
Hi All,
There is a parameter simultaneousMax=10 in oh323.conf.
Had anybody tried out what is the maximum value that can be achieved ?
What is the maximum number of simultaneous h323 calls can the oh323
driver can handle.
I tried to get it only till 30 to 40 simultaneous
What versions of OpenH323/Pwlib/asterisk-oh323 are you trying
to install?
Michael.
FaberK wrote:
Hello Guys,
first of all, I'm very new with asterisk.
I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7
Now I'm trying with asterisk-oh323
I've already installed pwlib, oh323
Mahmoud Badran wrote:
AVE!
i am trying to register h323 asterisk to the gatekeeper as i installed
asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323
on fedora core3 on a cisco mcs 7800 server problem is i want the
asterisk to register with gatekeeper endpoint with specific zone
Tony Mountifield wrote:
Hi, I'm using OH323, mostly with success, to interface Asterisk to
a provider's switch (World Telecom INX). I have noticed a particular
effect, and I wonder whether anyone else has seen the same?
The effect is audio flutter (almost like the flutter one gets on
MF or HF
Alistair Cunningham wrote:
Michael Manousos wrote:
Alistair Cunningham wrote:
I have a customer who wants to do large volumes of H.323 to H.323
hairpinning. We haven't tested this scenario for large volumes
before; maybe someone on asterisk-users has.
If they buy a top of the line PC, how many
Sebastian Atala wrote:
Hi,
Someone knows how can I register my Asterisk to a gatekeeper using
zone parameters?
I'm using asterisk 1.0.7 and oh323 0.6.5.
I'm trying to register to a gatekeeper in another network and I can't reach
this with a broadcast.
Zone is the name who Cisco call the GK
Jorge Alayon wrote:
Hi,
Does anybody knows how to konfigure oh323.conf to allow calls comming from a
peering gateway (i.e.: Cisco 5300) which is not connected to a gatekeeper,
and also from the gatekeeper to which Asterisk is registered ?
Nothing special here. Configure the channel driver with the
Hi Tony,
Can you get an ethereal trace of the RTP packets on both
directions? A short analysis of those streams (from within the
ethereal tools) would help us find the problem.
Michael.
Tony Mountifield wrote:
I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of
pwlib (v1.6.6.3) and
Try the 0.7.2-pre1 version of asterisk-oh323.
It can be found at the Download section on the home
page of asterisk-oh323.
Michael.
Jose R. Ortiz Ubarri wrote:
I have problems compiling the OH323 channel with Asterisk
CVS-HEAD-03/21/05-15:32:10.
I have the following errors.
chan_oh323.c:4895:
Tony Mountifield wrote:
Yesterday I wrote:
I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on
Fedora Core 3.
[... snip ...]
Well I gave up with chan_h323, which is a pity, because it should be the
solution that is better integrated with Asterisk. I would still like to
hear from
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Michael Manousos [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
When testing the ability of dual 3GHz Xeons to handle many simultaneous
OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323
is EXTREMELY profligate
Gabriel Millerd wrote:
I have been struggling with oh323 compilation for some time now. I am
trying to use the voip-info suggested walk through that points to here
...
http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en
... which asks for versions OpenH323 (v1.13.5)
Olle E. Johansson wrote:
During the developer's conference call yesterday evening,
it was decided that we finally should release the much-awaited
Asterisk 2.0 Stable release, also called codename AAFJ.
AAFJ as in Asterisk April Fool's Joke?
Nice :)
___
Cenk Yabas wrote:
Thanks to Yves's commitment I was able to configure oh323 channel, cleared
the codec issue, registered to Gatekeeper, placed a call, but receive this
message on the console. What might be the problem?
Asterisk Ready.
*CLI -- Registered with gatekeeper '[EMAIL PROTECTED]'.
I have prepared two new, not-final yet, releases of asterisk-oh323:
- 0.6.6-pre1 for Asterisk stable
- 0.7.2-pre1 for Asterisk CVS HEAD
They can be found at:
http://www.inaccessnetworks.com/projects/asterisk-oh323/download
Please try them and report problems at the bugtracker of
the channel driver
Hi Kamran,
Kamran Ahmad wrote:
hello
i was searching for solution to problem (sip-h.323).
any one from this list asterisk mailing have any idea
how to fix it.
i am getting error when i try to call from sip to
h.323 user
i am successfully registering my asterisk box with
gnugk. but when i try to
Hello all,
In an attempt to make easier and more effective the management of the
various issues/features/bugs of asterisk-oh323, I have setup a
bugtracker at:
https://skylab.inaccessnetworks.com/mantis
Please direct all the bug reports and contributed patches there.
Thanks,
Michael.
Kim Daeyong wrote:
I downloaded asterisk to use cvs to checkout the release version.
After installing, I would like to load module chan_h323.so but there is some
error :
*CLI load chan_h323.so
Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/m
odules/chan_h323.so:
Roger Schreiter wrote:
Michael Manousos schrieb:
...
Use Asterisk-1.0.3 with asterisk-oh323-0.6.5.
Hi,
may I ask, whether that combination runs really stable
at your machine?
I have now those versions installed.
I have asterisk crashes at least once every hour, when
several simultanious calls
asterisk-oh323-0.7.0 is for Asterisk CVS.
How did you manage to compile it with Asterisk-1.0.3?
Use Asterisk-1.0.3 with asterisk-oh323-0.6.5.
Michael.
Roger Schreiter wrote:
Hi,
which is currently a stable combination of asterisk and
asterisk-oh?
The combination of asterisk-1.0.3 and
Robert Rozman wrote:
Hi,
I have downloaded files and also local versions of pwlib oh323 (both Janus
patched). Both libraries compile fine, but I get following errors on
asterisk-oh323-0.6.5. Readme is a bit confusing since it doesn't mention
which local libraries should be downloaded from inaccess
Hi,
Enable the driver tracing (see wrapTrace* and libTrace* in oh323.conf),
re-run and send me the output file.
Michael.
Tola Ogunsan wrote:
Hi Michael and Everyone
I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm
getting this error
reason 24 (Call ended with Q.931 cause)
Hi Nicolas,
Andrew McRory has done some some job towards packaging asterisk-oh323.
The packages are available at:
ftp://ftp.linuxsys.com/pub/releases/
You could start with his packages and then move on.
Michael.
Nicolas FOURNIL wrote:
Hello
I'm trying for a while to compile and install OH323
Also the following has worked great for me:
http://www.wifive.net/introduction.asp
Michael
Radovan Mihalik wrote:
http://www.sjlabs.com/sjp.html
SJphoneR is a VOIP softphone that allows you to speak with any PC, PDA,
stand-alone IP-phone and with any legacy wired or mobile phone (using
your VOIP
Alexander Averyanov wrote:
Hello.
I'm try to set up asterisk for making outgoing calls with oh323 channel
driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37.
Our provider uses Mera MVTS softswitch and supports only H.323.
We don't use gatekeeper for connection but provider requires SOURCE
Adi Linden wrote:
I am experimenting with oh323 channels and h.323 gateways and a Cisco
CallManager. I am not using a gatekeeper at this time. Is it possible to
place calls coming into Asterisk from specific peers into specific
contexts?
Yes. You can associate called numbers/prefixes with contexts
The new configuration style of OH323 will simplify the sections of
the dialplan that handle H.323 calls.
Michael.
Roger Schreiter wrote:
Adi Linden schrieb:
...
In iax.conf eaxh peer has a context in which I can specify the
context an
inbound call will be placed in. I don't see anything
Silviu Herchi wrote:
Hello everybody,
Ive been pulling my hair for a week now over a problem, and I really dont
know where to look anymore. Heres my setup:
There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I
can use it to send and receive calls from physical phones attached
Silviu Herchi wrote:
Sorry, I mistakenly sent my mail before it was complete... Here it is again.
--
Subject: One-way audio in incoming calls with Asterisk + OpenGK +
Innovaphone IP3000
Hello everybody,
Ive been pulling my hair for a week now over a problem, and I really dont
know where to look
Hello all,
The new versions 0.7.1 (for Asterisk CVS HEAD) and 0.6.5 (for Asterisk
STABLE) fix a deadlock in outgoing H.323 calls and a bug that caused
chan_oh323 to update incorretly the DIALSTATUS variable.
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Hi,
Rafael J. Risco G.V. wrote:
Hello
I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4
and openh323-Janus_patch4 downloaded from inaccessnetworks so I did
this:
tar -zxvf openh323-Janus_patch4-src-tar.gz
cd openh323
patch -p1
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Francisco wrote:
Hi, im getting mad compiling the H323 channel (Jeremy's version
inAccess version). Ive tryed many versions of openh323 lib and pwlib,
and i get differets errors.
Does anyone uses this channel? and which version of it, openh323 lib and
pwlib?
asterisk-oh323-0.6.4 compiles/works
Tracy R Reed wrote:
Some of you may recall that I have been working on building a box to
convert H323 to SIP. After a significant amount of outside help and
slicing and dicing of the ohh323 code to get it to compile on AMD64 we
finally got it working. Now we are working on improving the
Daniel Eboa wrote:
Hello to all,
I have a strange behavior of my asterisk box. I'm running asterisk with
asterisk-oh323 channel driver and everything works very well.
But after few hours, my asterisk stop running and I have to restart it
by typing asterisk -vvvc. Most of the time I connect to my
-Original Message-
From: Michael Manousos [mailto:[EMAIL PROTECTED]
Sent: mercredi 1 décembre 2004 16:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours
Daniel Eboa wrote:
Hello to all,
I have a strange behavior of my
Daniel Eboa wrote:
How to get it?
Download it from here:
http://www.inaccessnetworks.com/projects/asterisk-oh323/download
-Original Message-
From: Michael Manousos [mailto:[EMAIL PROTECTED]
Sent: mercredi 1 décembre 2004 17:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Brian West wrote:
OH But it is just that simple.
You also have:
-= Info about application 'ImportVar' =-
[Synopsis]:
Set variable to value
[Description]:
ImportVar(#n=channel|variable): Sets variable n to variable as evaluated
on
the specified channel (instead of current). If prefixed with
Thanks.
I appreciate that.
Michael.
kido noagbodji wrote:
i have had some problems with the H323 channel ... Other party not
anwsering SIP 2 H323 bridge.
the chan_oh323 solves the problem. Use it.
(Even though it is quite complicated to install but READ the README file)
Nahuel that should solve
Roy Sigurd Karlsbakk wrote:
is it possible, from an agi script or directly in extensions.conf, to
override the DTMF and codec settings?
to answer my own question
SetVar(SIP_CODEC=g726)
allowed me to force g726, but only on outgoing calls.
when dialling in from the iax server, I do the same,
administrator tootai wrote:
Hi all,
I want to give a try to oh323 (currently nufone h323 channel is setup
and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib
1.8.1_ and _openh323 1.15.1_ What I made:
Wrong, wrong, wrong!
1) Read the README.
2) Get the right versions of
administrator tootai wrote:
Michael Manousos a crit :
administrator tootai wrote:
Hi all,
I want to give a try to oh323 (currently nufone h323 channel is setup
and compiling fine) on a yesterday CVS update of asterisk. I have
_pwlib 1.8.1_ and _openh323 1.15.1_ What I made:
Wrong, wrong, wrong
Al Escasa wrote:
In my oh323.conf, i am using:
userInputMode=TONE
Is everyone trying to say that i have no hope using
oh323 when using inband DTMFs? is this problem of
asterisk? the protocol? the codec? i wish there is
still some kind of workaround.. =(
What I meant was that inband DTMFs do not
Roy Layson wrote:
hope it can help
[deleted]
I. INSTALL OS
OS is Fedora Core2 installed only
-textbased internet (elinks)
-web server (apache and etc...)
-SQL (mysql and DBD/DBI)
-Development tools (default)
-kernnel
Tracy R Reed wrote:
On Tue, Nov 23, 2004 at 12:42:07PM +0100, administrator tootai spake thusly:
with my other regular x86 box running H323. One odd thing I note is that
when looking at the UDP traffic with tcpdump I see the * box receiving my
Same problem here. My * box is connected to GnuGK. CVS
Al Escasa wrote:
Hi everyone,
Could somebody enlighten me on this one? I have
configured my asterisk to run on oh323 using codec
g729. Incoming calls are working okay. But the thing I
want to work is say pressing some options, say dial 1
to go to voicemail or dial a certain number to dial a
Peter,
Peter Landy wrote:
New to Asterisk so I am sure this has been answered before. I can
compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323
then I get all kinds of errors even though I have set the paths up in
the source files. I can attach the errors if it is useful. I
kido noagbodji wrote:
Hello,
I just downloaded and installed the latest version of asterisk under
Fedora. (had it under FreeBSD but was having TOOO many problems)
After my installation i noticed that the channel H323 was not included (
I remember that i did not have to install it under
Try:
SetGlobalVar(OH323_OUTCODEC=g723.1)
Michael.
M. Ehsanul Karim wrote:
Hello,
What would be the outcodec value for g723.1 (6.3k). I have g723
support which works with SIP (not pass thru) , but when I use OH323 it
always
Unsupported ${OH323_OUTCODEC} value (G72316K3)!
I have enabled all g723
Since you are able to receive H.323 calls with chan_oh323, I assume
that the module is loaded. You could check the
incoming/outgoing/simultaneous limits or submit the oh323.conf.
Additionally, what are the full messages that you get on the
console?
Michael.
Alex van Es wrote:
Hi all,
For my setup
Alex van Es wrote:
Michael,
Yeah.. for sure the channel is loaded.. calling to my asterisks works fine.
I have included the oh323.conf and the original message.
Thanks a lot for you help. I would would like to get this baby working.
Alex
The log;
Nov 8 18:04:01 WARNING[294930]: channel.c:1901
Flash zap trunk application 0
app_zapbarge.so Barge in on Zap channel application 0
app_zapscan.so Scan Zap channels application 0
On 8-nov-04, at 18:29, Michael Manousos wrote:
Alex van Es wrote:
Michael,
Yeah.. for sure the channel is loaded.. calling to my asterisks
Hello all,
The asterisk-oh323 package has been updated. From now on, there are two
series of releases:
- 0.6.x releases, latest is 0.6.4. These will work with Asterisk v1-0
source code.
- 0.7.x and above, latest is 0.7.0. These are for CVS code of Asterisk.
Also, the latest versions now use
Sergio (RED) wrote:
Hi,
Anybody know if I can register my Asterisk in more than one h323 Gatekeeper.
I need to call to diferents providers depending on convenients
destinations prices.
This is purely an OpenH323 issue. The library does not permit such a
usage. I guess that Craig (Southeren) is
Huddleston, Robert wrote:
The only disadvantage we found to using the OH323 channel driver is that we
cannot now register netmeeting or other h323 directly to the * With the
What do you mean cannot now register? asterisk-oh323 doesn't implement
gatekeeper functionality. It never did. Just use
[EMAIL PROTECTED] wrote:
Hi,
I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each time
i run asterisk -c, i get the following error:
[chan_oh323.so] = (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/rtp.conf': Found
== Parsing '/etc/asterisk/oh323.conf': Found
Huddleston, Robert wrote:
Due to the format of the message coming from the H323 channels included w/
Asterisk we were unable to use our gatekeeper.
For a quick solution we tried the OH323 channel drivers and can receive
inbound calls from the parent gatekeeper.
We are trying to do a dial to
Huddleston, Robert wrote:
Okay - read it... my configuration works... what I want
exten = XX,1,Wait,2
exten = XX,2,Dial(OH323/XX)
I want it to pass the 10 digits to the DIAL string... I'm not sure I
understand the macros
can I just put the ${EXTEN} in there??
Of course. The
Rafael J. Risco G.V wrote:
Hello
Ive just install last cvs version (Mon Sep 6) of Asterisk with
asterisk-oh323-0.6.3b and pwlib-v1_6_6-src.tar.gz,
openh323-v1_13_5-src.tar.gz and .
this is the error loading asterisk with chan_oh323 module::
[cdr_csv.so] = (Comma Separated Values CDR
[EMAIL PROTECTED] wrote:
Hi there !
I searched the whole web to find some helping information about H323
Control Protocol, but there is no way to find that information.
We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2
+ 'asterisk-oh323_1.5 channel driver + wrapper' and
Joa~o Amaro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I Vlasis,
I'm using those versions (Fedora COre 1) and it compiled without
problems, but when i try to initialize asterisk i get the folowwing error:
ERROR [-1084337504]: chanoh323.c:4636 load_module: H.323 listener
creation failed.
It works fine for me on a Slack9.1 laptop.
Michael.
Vlasis Chatzistayrou wrote:
Hello,
I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2
installed but failed. I applied the patch to the required OpenH323 library
according to the instructions, and set the proper
Andres wrote:
The quick and dirty way:
In rtp.c, function ast_rtp_write, in the switch statement,
AST_FORMAT_G729A case, change the smoother creation to something
larger. E.g.:
rtp-smoother = ast_smoother_new(40);
Keep in mind that you must set this into something
Luis Vazquez wrote:
Does anybody knows if it's posible or if there is some develoment in
course to be able to use longer transmit packet sizes (as long as I know
this is fixed in 20ms now) with the compressed voip codecs in asterisk
(g729, g726, gsm, etc).
I need to use asterisk to connect
Enrico Stahn wrote:
Hi!
Have a look at the following entry. I solved this problem:
http://enrico.todo.de/weblog/item/asterisk-oh323-compile-error
That's the wrong way to do it. You use incorrect versions of
the libraries.
Michael.
___
Asterisk-Users
Kevin Walsh wrote:
[EMAIL PROTECTED] wrote:
On 27 Aug 2004 at 2:33, Kevin Walsh wrote:
There is no packet loss concealment in Asterisk at this time.
Why doesn't asterisk clock to the 1000 interrupts per second instead
of the incoming audio? Were there no interrupts available when it
started?
Kevin Walsh wrote:
Michael Manousos [EMAIL PROTECTED] wrote:
Look at the RTP stack of the receiver. When a packet is received, there
are two cases:
a) An RTP packet carrying voice frames is received. In that case the
decoder will play the voice frames.
b) A CN (Comfort Noise) packet is received
Roger Schreiter wrote:
Hi,
there are some posts about that topic, but
unfortunatelly I do not yet know what to do.
I find every call in Master.csv, but those coming in
via chan_oh323.
In oh323.conf I have
accountcode=oh323
but there is no other file in the directory cdr-csv
than Master.csv.
Can
Zineddin Karzazi wrote:
--- Robert Rozman [EMAIL PROTECTED] schrieb:
Hi,
I'd kindly ask for any guidance how to setup
Netmeeting to work with
Asterisk.
I've setup Asterisk as Gateway, selected GSM codec,
and I'm able to call
local extensions (no calls into PBX functions) but
get no sound.
Any
alex3377 wrote:
I' got a problem, using asterisk-rc2 :IVR functions
(Background...Playback...etc) doesn't works : Executing
Background(OH323/RX, vm-extension) in new stack
channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to
G729A---Asterisk box supplied only with network
Robert Rozman wrote:
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, August 26, 2004 4:52 PM
Subject: Re: [Asterisk-Users] RC2 and Netmeeting 3.01 ?
snip
There is no common
M. Willigs wrote:
Hi there.
I thy to compile asterisk-oh323-0.6.3a but it fail in the make command.
I have the pwlib-v1_6_6-1 and openh323-v1_13_5-1 as saying in the README
file of the packet asterisk-oh323-0.6.3a
You must apply the included OpenH323 patch before trying to
configure/compile
Chris A. Icide wrote:
I'm having a small issue with the oh323 implementation when it comes to
codec selection.
Version info:
CVS Head 6/30/2004
OH323 0.6.3
OpenPhone for windows version 1.8.1
Asterisk is configured as a h323 endpoint which either terminates to the
PSTN locally through a PRI or
lenz wrote:
Hello list,
as I'm writing a little perl parser for queue_log analysis, I'd like to
know *which* telephone answered a specific queue call. Unfortunately
app_queue only logs the call id but does not log the call end point.
This is okay for SIP endpoints, because their call id is
Zineddin Karzazi wrote:
Hi.
im compiling the wrapper for oh323(under Suse 9.0)
-pwlib 1.6.6
-openh323 1.13.5. (with oh323 Patch)
i execute:
./samples/simple/obj_linux_x86_r/simph323
and it works fine.
When i Run asterisk-oh323 0.6.2:
make
Download and install version 0.6.3a.
Steve Totaro wrote:
Does anyone do any large scale SIP to H323 conversion? How many
simultaneous calls can your server handle and on what hardware? I think
I read on the wiki that twenty five would max out most servers.
Not true for asterisk-oh323.
Micheal.
ruixun wu wrote:
Hi Alexey,
I followed your steps, but Asterisk still didn't
work. I am a little crazy. I show my envirement and
ld.so.conf here. Could somebody tell me if I am using
the correct libraries?
Thanks a lot
ld.so.conf:
/usr/kerberos/lib
/usr/X11R6/lib
/usr/lib/qt-3.1/lib
ruixun wu wrote:
Hi Michael,
Thanks for your time.
I deleted these two files
,libh323_linux_x86_r.so.1.13.5 and
libpt_linux_x86_r.so.1.6.6. And startd asterisk, the
error still exist.
Then I copy these two files from $PWLIBDIR/lib
and $OPENH323DIR/lib to /usr/local/lib. Startd the
There is nothing wrong with asterisk-oh323, the call is rejected from
the remote endpoint. Try to turn on only G.729 and retry.
And yes, you don't need g729 licenses to do g729 passthrough.
Michael.
David Allen wrote:
Hi All,
I have set up a box that will be used as follows:
SIP
Try to describe your problem. A first guess is that you didn't
apply the patch for the OpenH323.
Michael.
Mandar Pise wrote:
Hi Folks,
I am breaking my head for compiling asterisk-oh323 properly on my
asterisk box from past 1 week.
But still after my all efforts, I unable to make it compile
Kanuri, Seshu wrote:
I am wondering if anyone has a working install of oh323 on fedora Core2.
I'll try this when I find some time (I have to setup FC2 on a box).
You could help me by describing where it fails to install.
Michael.
An replies would be appreciated as we need this urgently.
Seshu
Why don't you use asterisk-oh323?
Michael.
Sebastian Nocetti wrote:
I WANT TO USE G729, I HAVE TO USE IT...
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Eric Wieling
Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m.
Para: [EMAIL PROTECTED]
Asunto: Re:
Hi Chris,
Chris A. Icide wrote:
According to the wiki at voip-info.org, the dial structure for using
oh323 without a gatekeeper is:
OH323/exten@host:port
or
OH323/exten
The second option is valid only in the case where a gatekeeper is used.
NOTE: OpenH323 library v1.12.0 has a bug in the
It seems that you use wrong versions of the libraries at
run-time (probably the distribution's libraries?). Do a
ldd /usr/lib/asterisk/modules/chan_oh323.so
Michael.
Fathallah Soumaya wrote:
when I put ldd /usr/local/lib/liboh323wrap.so, it
tells me:
libc.so.6 = /lib/tls/libc.so.6 (0x4200)
.
Michael.
Serge wrote:
Yes, it's work,
Thanks,
But possible don't use Global Var?, due in this situation all other
destinations use this codec, after 1 time global setup. And g729 - limited:(
Regards,
Serge.
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED
Try with 'SetGlobalVar' instead of 'SetVar'.
Michael.
Serge wrote:
Dear All,
I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but
it don't work.
oh323 driver don't want connect to gateway with g729, it's work if I
only use in oh323.conf one codec ( g729 ). If I enable 2 or
In oh323.conf set:
amaFlags=billing
Michael.
Oleg A. Arkhangelsky wrote:
Hello All,
It seems that this question is very stupid, but anyway. Do I need any
additional configuration for cdr_csv.so? This module is loaded by
default at Asterisk's startup (asterisk -fvvv):
[cdr_csv.so] = (Comma
Hi,
Do IAX(GSM) - IAX(ALAW) calls sound ok?
What is the configuration of OH323 channel (oh323.conf)?
Also, run asterisk with '-vvvcd', make a call and send the output.
Don't forget to enable the logging of debug messages (logger.conf).
Michael.
Arne Scheffer wrote:
Hello veryone,
I have a strange
What exactly is the problem with v0.6.3(a)?
Michael.
Anthony Law wrote:
I too tried 0.6.3 and it is behaving the same. I have now downloaded oh323
to 0.6.2a and it seems fine.
Regards,
Anthony
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Asterisk-Users mailing list
[EMAIL PROTECTED]
You are trying to compile an ancient asterisk-oh323 with fresh
Asterisk code. It won't work. Download and install
asterisk-oh323-0.6.3a. Also, download and compile the recommended
versions of OpenH323/Pwlib (OpenH323/Pwlib 1.12.2/1.5.2 are too old).
Michael.
mohammad mirzaee wrote:
HI ALL
HI
OK, I'll look at it.
Michael.
T. Chan wrote:
Dear All,
I don't know but I tried all 0.6.x version of OH323 and normally I use
safe_asterisk to start asterisk, and everytime when I use 'stop now' to
terminate asterisk, it does not do anything, and you are rite, I have to use
kill -9 to kill the
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