Hello List,
I've been trying to compile Asterisk with H.323 support and, after
correctly installing PTLib and H323plus (OpenH323), the Asterisk
configure script still doesn't detect the dependencies as installed.
I know they are correctly installed because after going into
Hi all,
I'm using asterisk 1.4.26.2.
I need to set TOS on H.323 channel.
Does chan_h323.conf support tos (or tos_audio) statement, as well as
sip.conf and iax.conf ?
Thanks,
Daniel
--
_
-- Bandwidth and Colocation Provided by
Hi All,
I would just like to clarify the requirements of the h323 channel within
asterisk.
Can I use a recent edition of PTLib and OpenH323, for example, the
editions located at OpenH323+:
http://www.h323plus.org/source/
OpenH323+ v1.20.2
PTLib v2.0.1
Or do I need to use the versions at the
://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister
Sent: Thursday, February 21, 2008 10:58 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] chan_h323 requirements
Hi All,
I
Hello,
To compile chan_h323 as is distributed you need to download OpenH323
v1.18.0 and PwLib v1.10.0 from:
http://www.voxgratia.org
Some months ago I had made a patch to compile the 1.4.x version and the
trunk version (which evolved to 1.6.x) with H323+.
Sadly, the patch was not included in
] On Behalf Of Bruce McAlister
Sent: Thursday, February 21, 2008 10:58 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] chan_h323 requirements
Hi All,
I would just like to clarify the requirements of the h323 channel within
asterisk.
Can I use a recent edition of PTLib
Hi,
Thank you for the details of which versions to get. I will be building
these two versions on Solaris to test chan_h323.
Did your patch for building with OpenH323+ make it into the 1.4 edition
of Asterisk?
Thanks
Bruce
Vlasis Hatzistavrou (KTI) wrote:
Hello,
To compile chan_h323 as is
Hello Bruce,
Bruce McAlister wrote:
Did your patch for building with OpenH323+ make it into the 1.4 edition
of Asterisk?
No, it didn't as it was considered a new feature and by Digium's policy
new features can only be added in the trunk versions.
The strange thing is that I added it in
I` using chan_h323 on my asterisk-1.4 to receive incomings calls. I need
to set just two codecs to receive this call (g723 and g729), but I`m using
disallow=all
allow=g729
allow=g723.1
In h323.conf, but when I received a call using codec g711 for example,
the call is answered, but doesn`t have
Hi All,
I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris
10. When I compile asterisk, the build fails at chan_h323 with:
--
chan_h323.c: In function `reload_config':
chan_h323.c:2863: error:
If I let modules.conf autoload chan_h323.so then when
I try to stop asterisk, it *does* stop (files in
/var/run/asterisk/ are removed and connection via -vr
from another console is not possible) but the
asterisk process stays alive and stalled. In other
words, a 'ps -ae | grep asterisk' show that
Dear Kiven;
Actually it is default and not degault. Also, I was
doing the compilation remotely via the Putty. Another
thing, I did another senario and got another thing, as
below:
I copied /usr/local/lib to /usr/lib and then I
restarted asterisk, but when I come back to run it,
then it was
Hi All;
I am trying now to compile h323 to be able to use it,
I did the pwlib and openh323 successfully and I
exported the PWLIBDIR=/usr/src/pwlib_v1_10_0 and the
OPENH323DIR=/usr/src/openh323_v1_18_0, then I was need
to compile h323 as following:
cd /usr/src/asterisk-1.4/channels/h323
When I
bilal ghayyad wrote:
cd /usr/src/asterisk-1.4/channels/h323
When I type make, it gives me:
make: Nothing to be done for 'degault'
This is *exactly* what showed up on your session? The word 'degault'
does not appear in the Makefile at all, so if that is the message that
you got then your
I have had some interesting compiling results with the
latest beta release of Asterisk. With reference to this channel
After running the make opt in the H323 directory, and the
make install in the Asterisk directory, there is still no chan_h323.so file
Created.. Are there any other
On Mon, Oct 23, 2006 at 04:09:43PM +0200, Patrick wrote:
I have had some interesting compiling results with the latest beta release
of Asterisk.. With reference to this channel.
After running the make opt in the H323 directory, and the make install in
the Asterisk directory, there is
PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chan_h323.so Asterisk Beta compilation
On Mon, Oct 23, 2006 at 04:09:43PM +0200, Patrick wrote:
I have had some interesting compiling results with the latest beta
release of Asterisk.. With reference to this channel
Ganbold Tsagaankhuu wrote:
Hello,
I installed Asterisk from CVS on Redhat Linux 9 and working with
chan_h323 module and g729/g723 free codecs too.
My network connection diagram:
--
X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN
boldsoft*CLI
: [Asterisk-Users] chan_h323 problem
Ganbold Tsagaankhuu wrote:
Hello,
I installed Asterisk from CVS on Redhat Linux 9 and working with
chan_h323 module and g729/g723 free codecs too.
My network connection diagram:
--
X-lite/X-Pro--Asterisk--chan_h323
Hello,
I installed Asterisk fromCVSon Redhat
Linux 9 and working with chan_h323 module and g729/g723 free codecstoo.
My network connection diagram:
--
X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN
boldsoft*CLI show versionAsterisk
Hello,
I installed Asterisk from CVS on Redhat Linux 9 and working with
chan_h323 module and g729/g723 free codecs too.
My network connection diagram:
--
X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN
boldsoft*CLI show version
Asterisk
I'm attempting to get chan_h323 working on Asterisk CVS-STABLE.
I've compiled it ok using the Janus release of pwlib/openh323, by
editing the makefile as per the comments.
Call setup and cleardown seems to work fine, but no audio is being
passed in either direction.
Doing an h.323 trace 9, I
Please post ur installation script for chan_h323
- Original Message -
From: Atif Rasheed [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, June 20, 2005 7:21 AM
Subject: [Asterisk-Users] chan_h323 vs chan_oh323 chan_ooh323
hello there,
can somebody please
hello there,
can somebody please comment which one of these channel drivers will give
best output doing g729|g723 pass-thru. only pass-thru is needed no
transcoding.
please share your experience. if somebody has some figures (simultanous
calls using a certain channel driver) it will be
Hi all,
All incoming H.323 calls on chan_h323 were forwarded to default
context but not detroit. It seems context=detroit is not effective.
Any helps???
[det-gw]
type=h323
prefix=1248,1313
context=detroit
Thanks.
IM
___
Asterisk-Users mailing list
Hi
I've installed successfully:
- PWlib v1.6.7 library
-Openh323 v1.13.5 library
-asterisk-oh323 v0.6.5
and so the modules chan_oh323 is installed successfully
Now I try to install chan_h323
First question: is this necessary?
I edit the Makefile in the directory
[EMAIL PROTECTED] a écrit :
Hi
I've installed successfully:
- PWlib v1.6.7 library
-Openh323 v1.13.5 library
-asterisk-oh323 v0.6.5
and so the modules chan_oh323 is installed successfully
Now I try to install chan_h323
First question: is this necessary?
No, it's or oh323 or h323. I suggest
.
Regards,
Nir S
- Original Message -
From: Chetan Sarva [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 04, 2005 11:08 PM
Subject: [Asterisk-Users] chan_h323 codecs
Hi,
Can anyone confirm that if I want to do h323 proxying that I do not need
codecs installed
Hi,
Can anyone confirm that if I want to do h323 proxying that I do not need
codecs installed? For example if the codec being used is g723.1, I don't
need the codec installed locally because there is no compression or
decompression being done on my server; the incoming traffic is simply
being
Ok, I have some more info. The code from openh323.org will not compile on
x86_64 but the latest from the OpenH323 project on sourceforge will
compile just fine on x86_64. Asterisk 1.0 will not compile with this new
openh323 code but it looks like the latest cvs-head does. In
channels/chan_h323 I
Tracy R Reed a écrit :
[...]
bit64*CLI show channels
Channel (ContextExtensionPri ) State Appl. Data
SIP/pstn1-e7c5 (default 1 ) Up Bridged Call H323/ip$[myip]:30005/28852
1 active channel(s)
Nov 24 02:51:14 WARNING[10527]: channel.c:494
Tracy R Reed a écrit :
Has anyone here done this? I got it compiled just fine but when I make a
call I do not get any audio going either way. The * box is not behind any
sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I
have it set up properly to work through NAT and it
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 Head 11/02/04,
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
Michael Manousos a écrit :
Tracy R Reed wrote:
[...]
Same problem as in you ran tcpdump or something and saw the odd behavior
of receiving but not sending any packets? VERY interesting. Were you
on an
x86-64 bit box or regular x86? I was thinking this odd behavior was some
odd interation with
Has anyone here done this? I got it compiled just fine but when I make a
call I do not get any audio going either way. The * box is not behind any
sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I
have it set up properly to work through NAT and it will talk correctly
with
Hi All
Is there a better mailing list where I should ask these questions ?
Thanks
Mike O'Connor wrote:
Hi all
I spent a few hours trying to information on asterisk, h323 and sip
support for codecs with 20ms packetisation, and have not been able to
find anything relivatant.
Our supplier of call
Is this mike oconnor as in the Australian mick oconnor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
O'Connor
Sent: Wednesday, 20 October 2004 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
HI Mike,
You wouldn't be trying to connect to Comindico in Australia by any
chance?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike O'Connor
Sent: Monday, 18 October 2004 02:05
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] chan_h323
Hi all
I spent a few hours trying to information on asterisk, h323 and sip support for codecs
with 20ms packetisation, and have not been able to find anything relivatant.
Our supplier of call termination requires h323 the following:
* The signalling port is 1720
* H.323 version 2 with fast start
Hi,
I'm looking for a mean in chan_h323 to jump to a specific
context dependent on the remote ip address.
E.g. an argument, let's tell it ignore_h323_name, in h323.conf
users like this:
[BillyBob]
ignore_h323_name=yes
type=user
host=1.2.3.4
context=path1
in a way, every incoming call from ip
Hi,
I have the following topology:
PSTN/H323 gateway-GNUGK-chan_h323/chan_sip-SIP EP
Mostly everything works fine except chan_h323 is not passing
audio from PSTN before the call is answered and as a result users
can't hear PSTN announcements (like the number is not in service)
that's played on
I tried a lot of
times to get it worked, but I cant obtain audio using SIP-chan_h323 or
chan_h323-SIP
I tried disbling
FastStart without good results...
What's the
problem?
I need to do BRIDGE
between SIP and H.323!!
help!!
Sebastian.-
: [Asterisk-Users] chan_h323 no audio both ways
I've compiled chan_h323 with the latest cvs code, but my calls don't
pass audio.
The call connects just fine, as there are no errors reported on either
side, nor in a traffic examination with ethereal.
I've tried the following:
voip
-Users] chan_h323 no audio both ways
I've compiled chan_h323 with the latest cvs code, but my calls don't
pass audio.
The call connects just fine, as there are no errors reported on either
side, nor in a traffic examination with ethereal.
I've tried the following:
voip phone - asterisk - asterisk
I've compiled chan_h323 with the latest cvs code, but my calls don't
pass audio.
The call connects just fine, as there are no errors reported on either
side, nor in a traffic examination with ethereal.
I've tried the following:
voip phone - asterisk - asterisk - voip phone
voip phone -
Hello,
Has anyone gotten chan_h323 and
chan_oh323 to run on the same system at the same time? Provided you change the
listening ports of course. I can get both of them to start, but whenever I try
to make a call using chan_h323 I get a segmentation fault. This doesn't happen
if I disable
Title: Chan_h323 docs
Jeremy,
In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution.
Could you post those docs in your download directory?
I'm trying to understand the nuances of your driver, gnugk, and
Ok, I've managed to get inbound and outbound calling to work with chan_h323
and gnugk.
A few questions:
1) if I do a reload in *, chan_h323 loses its registration with gnugk, and
will no longer pass calls to it. A second reload will crash *. Is this
supposed to be?
2) For a configuration in
PROTECTED]
To: [EMAIL PROTECTED] Digium. Com [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_h323 readme file
Reply-To: [EMAIL PROTECTED]
If you have problems use the latest ones. Otherwise use whats listed
because thats what JJ has tested and is known to work.
bkw
On Mon, 8 Dec 2003, SW
Hello
I am getting ready to install chan_h323. Just updated my * with the latest
code from CVS (12/08/03). I was reading the Readme file and confused.
Quoted from the README
NOTICE: Whatever you do, DO NOT USE distrubution specific installs
of Open H.323 and PWLib. In fact you should check to
If you have problems use the latest ones. Otherwise use whats listed
because thats what JJ has tested and is known to work.
bkw
On Mon, 8 Dec 2003, SW wrote:
Hello
I am getting ready to install chan_h323. Just updated my * with the latest
code from CVS (12/08/03). I was reading the Readme
If anybody still have any G729 handshake problem with Asterisk and other
non-Digium partner, I *really* recommend to use this patch:
http://bugs.digium.com/bug_view_page.php?bug_id=421
6 monhts passed and finally my problem seems to be solved.
Thanks Adam!
Isamar
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0
Are you using the recommended pwlib and openh323 tarballs?
bkw
On Mon, 13 Oct 2003, CW_ASN wrote:
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used
gatekeeper = DISABLE
[Gustavo]
type=user
host=10.60.144.14
context=default
incominglimit=31
Regards,
Gus
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 13, 2003 4:46 PM
Subject: Re: [Asterisk-Users] chan_h323 - Segmentation fault (core
We have an odd problem, where inbound H323 (chan_h323) calls will sometimes
cause a Ringing Congestion that appears to keep the channels open and never
release it until we kill and restart asterisk. These Ringing Congestions
start to pile up, which eventually crashes Asterisk.
H323 Gateway -
Hello,
Camparing chan_h323 config with chan_oh323 config,
In the codec section chan_oh323 allow me to specify frame value.
Is there a equivalent in chan_h323? Or if not, what
is the default frame value if I use G.729(digium).
Foong
Since then I couldn't test it, but now I installed EtheReal last
version with h323 support. Did some calls and perceived that the
call is being cut after the Master/Slave negotiation.
Asterisk is sending an EndSession as you can see in the file
attached.
If the list doesn't allow attachments, the
hi
IIRC, Jeremy once said that chan_h323 could be used as a gatekeeper but
perhaps lacking a few features as compared to gnugk. Is this possible? I
have some dlink DPH-100H phoes here for testing, but they require a
gatekeeper, and if I can do it, I'd love to keep gnugk out of this.
thanks
roy
This happens only on relaod. You can disable reload routine in chan_h323.c
...
Martin
On 1 Sep 2003, Michael wrote:
I'm running the CVS from last week and from day one (over 4 months now)
I've had this problem where asterisk core dumps when using chan_h323.
It appears to be a problem with
On Tue, 2003-09-02 at 09:24, Martin Pycko wrote:
This happens only on relaod. You can disable reload routine in chan_h323.c
...
Thanks. I'll give it a try.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
It only core's when u use the gatekeeper component, due to the way pwlib
deals with memory allocation. This is going to take quite a lot of
trying various different incantations to fix, unfortunately I cannot
justify dedicating that kind time, at this point.
Sorry,
Jeremy McNamara
Martin
I'm running the CVS from last week and from day one (over 4 months now)
I've had this problem where asterisk core dumps when using chan_h323.
It appears to be a problem with pwlib and the console, but I'm not sure
how to read the below output from gdb. I can start Asterisk just fine
and chan_h323
On Mon, 2003-09-01 at 11:19, Brian West wrote:
Are you using the recommended pwlib and openh323 versions?
Yes.
lrwxrwxrwx1 root root 12 Aug 17 20:39 pwlib -
pwlib-1.4.11
lrwxrwxrwx1 root root 15 Aug 17 20:01 openh323 -
openh323-1.11.7
Michael
Does chan_h323 support phone number calling via a gateway? ie.,
something like calling 5000 forwarded to:
exten = 5000,1,Dial(h323/[EMAIL PROTECTED])
if so - what format should the exten be in? Thanks.
Regards,
Steven Thomas
___
Continuing my problems with h323. I think I am getting closer.
SJPhone works direct to the gateway - calls and answers fine on the pstn.
So the gateway is working.
Inbound calls from PSTN = Gateway = Asterisk = Phone work great!
Outbound from Asterisk = Gateway = PSTN still remains a
I have on Chan_h323 with G729 and X100P trying to connect to
a Planet VOIP400 gateway box(http://www.planet.com.tw)
I uncommented g729 in the Makefile and I'm setting g729 in h323.conf
I'm receving in my side:
1:20.906 H225 Caller:810f070 h323ep.cxx(1537)
H323 Clearing
Hi
The endpoint seems to be running Radvision h323 stack, and I know
chan_h323 works with Radvision, there could be a couple of reasons!!
1) You dont have G729A in the capabilities of remote endpoint
2) The packetization interval is way off
The best way would be to run ethereal or dump323 and
Well depends.. what kind of problem are you having?
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_problem_troubleshooting09186a00800c5e33.shtml
Check those... I suspect one of those has nailed
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no
.
- Original Message -
From: Sean Figgins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 12:22 PM
Subject: Re: [Asterisk-Users] chan_h323.c
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install
PROTECTED]
Sent: Wednesday, August 20, 2003 12:22 PM
Subject: Re: [Asterisk-Users] chan_h323.c
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote
What happened to chan_h323.c in the asterisk
cvs? I got ast_h323.cpp, ast_h323.h and chan_h323.h but no
chan_h323.c. Hence chan_h323.so was not created so no h323 support in
asterisk.
Just wondering when to expect it again because I
was stupid and didn't make a backup of the asterisk code
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
Mark
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp, ast_h323.h
and chan_h323.h but no chan_h323.c. Hence chan_h323.so was not
Hi,
I have been using chan_oh323 with a latency issue even on the same network.
I am now trying chan_h323 and can only get one way audio. I am testing
using SJPhone - SJPhone, and also SJPhone - 7960 (SIP).
Any ideas? Must be something obvious that I am missing?
Thanks.
Regards,
Message -
From: Steven Thomas [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 17, 2003 8:56 PM
Subject: [Asterisk-Users] Chan_h323 one way audio
Hi,
I have been using chan_oh323 with a latency issue even on the same
network.
I am now trying chan_h323 and can only get one way
:
[EMAIL PROTECTED]Subject: Re: [Asterisk-Users]
Chan_h323 one way audio
.digium.com
]
To: [EMAIL PROTECTED]
Sent: Monday, August 18, 2003 11:30 AM
Subject: Re: [Asterisk-Users] Chan_h323 one way audio
not sure what you mean by 'are you running cvs'?
What does the TOS setting do?
Regards,
Steven Thomas
Kelvin Chua
Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or
do you have to patch or add it in to the source directory structure before
compiling?
Can / and maybe how can this be added after?
Thanks.
Regards,
Steven Thomas
___
Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or
do you have to patch or add it in to the source directory structure before
compiling?
Can / and maybe how can this be added after?
H.323 is coming into asterisk cvs, and i think is trying to find if you
have openh323,
We use rfc2833 for everything and have no trouble. Make sure your 7960
is sending the right indications.
Jeremy McNamara
Jay Sakata wrote:
I have the same problem that Michael describes below does anyone have any recommendations?
Jay
] On Behalf Of Jeremy
McNamara
Sent: Tuesday, August 12, 2003 11:53 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_h323, Asterisk and DTMF issue
We use rfc2833 for everything and have no trouble. Make sure your 7960
is sending the right indications.
Jeremy McNamara
Jay Sakata wrote:
I
Having problems to connect another device using chan_h323.
When G723.1 or G711: log says:
NOTICE[15376]: File channel.c, Line 1325 (ast_set_read_format): Unable to
find a path from 1 to 64
NOTICE[15376]: File channel.c, Line 1296 (ast_set_write_format): Unable to
find a path from 4 to 1
be inline .. THe (thousands of) error messages
aren't really a problem, just annoying.
Dave
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 15, 2003 4:28 PM
Subject: Re: [Asterisk-Users] Chan_H323, G729 (minor problem)
You're trying
]
Sent: Tuesday, July 15, 2003 4:28 PM
Subject: Re: [Asterisk-Users] Chan_H323, G729 (minor problem)
You're trying to detect inband dtmfs from the codec stream.
Martin
On Tue, 15 Jul 2003, Dave Alan Caruana wrote:
hi ..
I have finally managed to get Chan_H323 G729 working
hi ..
I have finally managed to get Chan_H323 G729 working
flawlessly, thanks to some help from Jerry McNamara.
For those out there who are stuck with the same problem
the procedure is :
1. install on RedHat 8.0 and nothing else (RH9 doesn't work!)
2. Install asterisk, zaptel etc. the normal way
You're trying to detect inband dtmfs from the codec stream.
Martin
On Tue, 15 Jul 2003, Dave Alan Caruana wrote:
hi ..
I have finally managed to get Chan_H323 G729 working
flawlessly, thanks to some help from Jerry McNamara.
For those out there who are stuck with the same problem
the
Hi folks,
Im using chan_h323 to dial out to a gateway which connects me to the PSTN.
In order to use a menu system such my bank menu system, I have to set
dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info
wont work with Asterisks voicemail system.
Im using the
Hello all,
I got the following error compiling h323 support in the latest cvs. Below
the error is a diff to the file that I got to make it work. I took an
example out of sip as far as the syntax for ast_rtp_new. Not sure if it is
correct or not, but it seems to work. Please correct me if I am
I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled,
and chan_h323 module does not load with undefined symbol
_ZTI19H323AudioCapability. What could be the problem?
Peter
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
This is covered in asterisk/channels/h323/README
RTFM
Jeremy McNamara
Peter Zeltins wrote:
I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled,
and chan_h323 module does not load with undefined symbol
_ZTI19H323AudioCapability. What could be the problem?
Peter
I'm having a problem with chan_h323 compiling for Asterisk.
RedHat 7.3
PWLIB 1.4.11 pwlib_1.4.11.tar.gz
OpenH323 1.11.7 openh323_1.11.7.tar.gz
[EMAIL PROTECTED] h323]# make clean
rm -f *.o *.so core.*
[EMAIL PROTECTED] h323]# make
cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC
you need to build pwlib and/or setup your environment properly.
See asterisk/channels/h323/README
Jeremy McNamara
[EMAIL PROTECTED] wrote:
I'm having a problem with chan_h323 compiling for Asterisk.
RedHat 7.3
PWLIB 1.4.11 pwlib_1.4.11.tar.gz
OpenH323 1.11.7 openh323_1.11.7.tar.gz
[EMAIL
PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, June 16, 2003 11:27 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] chan_h323 - pwlib 1.4.11, openh 1.11.7 comiple
problems
I'm having a problem with chan_h323 compiling for Asterisk.
RedHat 7.3
PWLIB 1.4.11 pwlib_1.4.11.tar.gz
OpenH323 1.11.7
on a clean RedHat 7.2 100% install
I hope something in there helps...
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 16, 2003 8:20 PM
Subject: RE: [Asterisk-Users] chan_h323 problems
I did RTFM. It looks like the instructions conflict each other
Hi,
trying to build the h323 channel driver that comes with asterisk works
fine, but only as long as I use openh323-1.11.7.
Unfortunately, that setup seems to have a bug which misguides one of the
audio streams. (So while * can hear me, the phone remains silent.)
I suppose that bug is fixed at
Hello,
I've been working with the chan_h323 myself, and I had several problems,
but finally got it working.
I had to do things in the following order:
(1) build and installed asterisk as root
(2)I built pwlib and openh323 into my home directory (not root) and
built them there as me, I
If you would have followed the build instructions laid out by the Open
H.323 folks you wouldn't have had to go thru all of that.
http://www.openh323.org/build.html
(Notice they NEVER tell you to make install ANYTHING, there is a reason
for that)
Jeremy McNamara
Kelly McDonald wrote:
On Tue, 10 Jun 2003, Jeremy McNamara wrote:
trying to build the h323 channel driver that comes with asterisk works
fine, but only as long as I use openh323-1.11.7.
Unfortunately, that setup seems to have a bug which misguides one of the
audio streams. (So while * can hear me, the phone
1 - 100 of 110 matches
Mail list logo
