-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, December 07, 2006 5:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] oh323.conf question
Hi all,
I would like to know if it exists the possibility to
Hello Marcus,
Check your gatekeeper.ini
**
TotalBandwidth=
**
Don't Forget a gatekeeper use RAS to manage the
bandwidth .
try to add bandwidth !
Harry
PS:
Do you want to test your h323 terminal with my
gnugk/asterisk ?
--- Marcus Carlson [EMAIL PROTECTED]
Hi Harry,
Tried setting the bandwidth to different sizes (from 128 up to 1024000)
and both tried my normal account and the special SwyxGate account. Same
result.
I would very much appreciate if I could try against your H323 server.
Marcus
If you want realtime chat you could reach me at
[EMAIL PROTECTED] wrote:
Hi all, I installed asterisk 1.2 branch, with oh323 channel support.
Everything is fine, with netmeeting I can call and receive incoming calls,
internal and external
Then I tried to setup an AT320 phone , which is based on PA168S chip.
Which version of the
/2006 11.42 Re: [Asterisk-Users] OH323 issue on
AT320 Phones
Please respond
I have installed oh323 channel driver (finaly! :)). I head some problem
starting * so I have put the smallest possible oh323.conf file to se what
happens. When I don't put available codec's in oh323.conf (*1) Asterisk
starts but he also disables h323 channel because there are no available
Hello
We have detected that the newer Cisco IOS versions include a
SignalUpdate message after each alphanumeric UserInputIndication.
Did the oh323 asterisk module support SignalUpdate?
Has anybody know something?
Thanks
Jsalas
___
--Bandwidth and
An easy way to do that, if you do not neet to register on a gkp, its
doing a dial OH323/ipgateway:port
Did you try this?
Abdul Lateef escribió:
Hi all,
I have H.323 Gateway, and i want to make a peer to
route calls to this GW. But i don't know is oh323.conf
supporting to add peer type entry
Hi,
i treid this
OH323/ipgateway:port
and working well for me. But i need to add some more
featurres, like some of my H323 GW supporting only
G.7231 codec and some one G.729 and others feature
like rtptimeout etc
So if i am direct dialing without these feautres, the
GW are not able to
Yes, I got similar error the last time I tried.
On 1/24/06, Juan Salas [EMAIL PROTECTED] wrote:
Hello all.
Has anybody work with asterisk version 1.2.2 and
oh323 module?
When I try to make it show:
wrapendpoint.cxx:800: error: expected primary-expression before ')' token
Sent: Thursday, December 29, 2005 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] oh323 configuration
El jue, 29-12-2005 a las 05:40 +0500, Rehan Ahmed escribió:
Hi,
What exactly would you like to do, how would you like asterisk to talk
Sorry... the question is related with ooh323
It's possible to register ooh323 with gnugk ?
Any on knows one good ooh323 how to?
On Wed, 2005-12-28 at 09:48 -0500, Guillermo Salas M wrote:
It's possible to register oh323 with gnugk ?
Any one knows one good oh323 how to?
Regards,
--
Hi,
What exactly would you like to do, how would you like asterisk to talk with GNUGK
Rehan
On 12/28/05, Guillermo Salas M [EMAIL PROTECTED] wrote:
It's possible to register oh323 with gnugk ?Any one knows one good oh323 how to?Regards,
--Guillermo Salas M.Telconet S.A. MantaCalle 15 y Av. 24
El jue, 29-12-2005 a las 05:40 +0500, Rehan Ahmed escribió:
Hi,
What exactly would you like to do, how would you like asterisk to talk
with GNUGK
I'm a little confused about the use of ooh323. I want to register some
elesign h.323 hardare with gnugk to call to sip devices conected with
You should have more info in full log messages, look to this file and send
output.
Adam
Cytowanie Rafael R. GV [EMAIL PROTECTED]:
Hello
I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk
1.2libraries, must be
oh323-0.7.3, now I have compiled this version but when reload
/var/log/asterisk/full.1 output:
Nov 26 21:25:39 VERBOSE[14215] logger.c: [chan_oh323.so]Nov 26
21:25:39 WARNING[14215] loader.c: /usr/lib/asterisk/modules/chan_oh3
23.so: undefined symbol: _ZNK8PChannel7IsClassEPKc
Nov 26 21:25:39 WARNING[14215] loader.c: Loading module chan_oh323.so failed!
It looks like compiling oh323 with wrong version of headers or wrong version of
open323/pwlib. Are you completly sure that you deleted old headers and
libraries when upgraded asterisk to new version?
Adam Rybak
Cytowanie Rafael R. GV [EMAIL PROTECTED]:
/var/log/asterisk/full.1 output:
Nov
@lists.digium.com
Sent: Sunday, November 27, 2005 4:21 PM
Subject: Re: [Asterisk-Users] OH323 channel in asterisk 1.2 ...Ouch ...error
while writing audio data!
It looks like compiling oh323 with wrong version of headers or wrong
version of
open323/pwlib. Are you completly sure that you deleted old
[EMAIL PROTECTED] wrote:
WARNING[12461]: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK8PChannel7IsClassEPKc
Oct 10 14:29:20 WARNING[12461]: Loading module chan_oh323.so failed!
Does anybody know which is the problem ?
It seems Asterisk source and binary version do not fit.
After more test, get the following:
Oh323's call-id works on asterisk1.07+ oh323 (0.66), doesn't work
on ( asterisk1.2.0beta1+oh323 0.73), any pathch to get oh323 0.73
works?
On 10/2/05, Asterisk guy [EMAIL PROTECTED] wrote:
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73),
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73),
any suggestion for this ?
On 6/13/05, Federico Alves [EMAIL PROTECTED] wrote:
I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I
set the caller id correctly in my perl AGI script
$AGI-set_callerid($ani); , the
I have the same problem with asterisk CVS HEAD and asterisk-ooh323 module from
obj-sys. This driver is included into asterisk-addon package.
Adam.
Cytowanie Asterisk guy [EMAIL PROTECTED]:
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73),
any suggestion for this ?
On 6/13/05,
I have the same problem with asterisk CVS HEAD and asterisk-ooh323 module from
obj-sys. This driver is included into asterisk-addon package.
Adam.
Cytowanie Asterisk guy [EMAIL PROTECTED]:
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73),
any suggestion for this ?
On 6/13/05,
On Wed, Sep 28, 2005 at 06:33:23AM +0100, Ade Agbero wrote:
I have successfully installed oH323, but when I make a call the recipient can
hear me but I can't hear anything at my end, so voice is only travelling in
one direction.
Client Softphone (SIP) = Asterisk = H323 Gateway.
All
I found out Voice in one direction only is a bug within oH323, which has been corrected in Version 0.6.1 (see http://www.inaccessnetworks.com/projects/asterisk-oh323)
However, I am using [EMAIL PROTECTED], I don't know if I can install the latest oH323 version on [EMAIL PROTECTED], if so, which
I would try 0.6.7, since [EMAIL PROTECTED] is based off of the stable 1.0.x branch of Asterisk (excluding the fairly recently released 1.2Beta version of [EMAIL PROTECTED], which I assume you are not using.)What version of [EMAIL PROTECTED] are you using?TomOn Sep 28, 2005, at 1:18 PM, Ade Agbero
Subject: Re: [Asterisk-Users] oH323 Voice
in one direction only
I would try 0.6.7, since [EMAIL PROTECTED] is based
off of the stable 1.0.x branch of Asterisk (excluding the fairly recently
released 1.2Beta version of [EMAIL PROTECTED], which I assume you
are not using.)
What version
List - Non-Commercial Discussion
Sent: Wednesday, September 28, 2005 12:11 PM
Subject: Re: [Asterisk-Users] oH323 Voice in one direction only
I would try 0.6.7, since [EMAIL PROTECTED] is based off of the stable 1.0.x branch of Asterisk (excluding the fairly recently released 1.2Beta version of [EMAIL
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
Sahil Gupta wrote:
Client (MERA) -- H323 -- Asterisk -- IAX -- Asterisk
You don't specify which H.323 channel driver you are using; there are
least four possibilities at this time, so that would be helpful information.
___
--Bandwidth and
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
, so better ask
the provider.
LTenorio
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Mehdi chouikh
Sent: Thursday, September 01, 2005 12:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
Ducat [mailto:[EMAIL PROTECTED]
Sent: Friday, September 02, 2005 5:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] oh323 or h323
LTenorio,
I am not sure what you mean between Terminal and Gateway.
The voip providor in China
.
LTenorio
-Original Message-
From: Steve Ducat [mailto:[EMAIL PROTECTED]
Sent: Friday, September 02, 2005 5:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] oh323 or h323
LTenorio,
I am not sure what you mean
: Friday, September 02, 2005 9:54 AM
To: Leandro Tenorio
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] oh323 or h323
LTenorio,
Then this is my problem. I can only register to the gatekeeper as a
terminal, they do not allow me to register as a gateway
OK, now I am getting closer.
I am trying to get asterisk to connect to a gatekeeper as a terminal
to receive calls made to the landline number in china which is passed
along via h323.
I have now concentrated on only 1 number.
Protocol = H323
Gatekeeper = 210.21.118.xxx
H323ID =
Hello
Personaly i prefer oh323, i am using for one year whitout problems.
and is more easier to configure.
regards
On 9/1/05, Steve Ducat [EMAIL PROTECTED] wrote:
I have just signed up for 2 landline numbers in China. They haveoffered to sell me 2 h323 compatible handsets which I have declined
] On Behalf Of Mehdi
chouikhSent: Thursday, September 01, 2005 12:11 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] oh323 or h323
Hello
Personaly i prefer oh323, i am using for one year whitout problems.
and is more easier to configure.
regards
On 9/1/05
Abdel Rahman Tarzi wrote:
Following the install, I do not seem to have the option to create an
oh323 extension or trunk. Something that I need to do.
I realize it’s possible to edit the .conf but I needed to ask whether
this was “normal” – doesn’t seem like it is to me.
Naturally, I’m
Abdel Rahman Tarzi wrote:
I installed oh323 and everything seemed to go smoothly (compile
everything upto calling through using oh323).
I must admit, there is some behavior that’s doesn’t seem right but
generally, I’m able to dial-out of any oh323 device whether to an
extension or to a
On Mon, August 22, 2005 20:21, CM Rahman Jr. said:
Anybody here using iax2 for one call leg and other call leg for oh323? I
am
getting broken sounds from Iax2 call get.
Can somebody here help?
Thanks
OT: What does this have to do with the small office / analog thread?
Anyway:
H.323 isn't
several factors :
- check 'show translation' from asterisk to see how long it will take
for transcoding between your codecs. with your machine, should not be
long.
- the h323 endpoints latency. ( a lot of times this attributes to delay.)
- echo cancellation and zitter buffer ( zitter
On 27-07-2005 at 03:38:19PM -0700, Bashir Ullah wrote:
Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8
, fedora 1 , asterisk 1.0.6Hi All
I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb
ram, with g729 for i686 , (fedora 1).
my
by quintunm
codec=G72316K3
frames=24
- Original Message -
From: Apu Islam [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 28, 2005 12:14 AM
Subject: Re: [Asterisk-Users] oh323 geting voice problem
On Sat, 2005-07-09 at 17:59 -0500, CM Rahman Jr. wrote:
Hi,
I have downloaded
asterisk-oh323-0.6.6.tar
pwlib-Janus_patch4-src-tar
openh323-Janus_patch4-src-tar
pwlib and openh323 compiled fine as instructed.
When I tried to compile asterisk-oh323
Try this link:
@lists.digium.com
Sent: Monday, June 20, 2005 1:11 AM
Subject: RE: [Asterisk-Users] OH323 with g723
Hi,
Visit http://aussievoip.com.au/wiki-G723-1-Install you'll find how to
install g723, but first you have to install g729
http://aussievoip.com.au/wiki-G729-Install
I have tested
Hi,
Visit http://aussievoip.com.au/wiki-G723-1-Install you'll find how to
install g723, but first you have to install g729
http://aussievoip.com.au/wiki-G729-Install
I have tested it with Quintum, it works
Enjoy :)
Erdem HAKI - [EMAIL PROTECTED]
-Original Message-
From: [EMAIL
ok. I've worked out that G.711 is 1ms of audio per frame... what about G.729?
Rod Bacon wrote:
Forgive this (possibly) silly question, but my upstream provider
requires a packetization of 20ms. Using asterisk-oh323, I can set the
number of frames per RTP packet. How does this equate to
I answered my own silly question.
10ms.
If anyone needs a working OH323 config for Comindico (SPT) in Australia, please
mail me (G.729 and G.711).
Rod Bacon wrote:
ok. I've worked out that G.711 is 1ms of audio per frame... what about
G.729?
Rod Bacon wrote:
Forgive this (possibly)
Hi jeromy,
They are codec's problem?
How are you configured the files:
extensions.conf
oh323.conf
sip.conf
??
Rafael
Mensaje citado por Jeromy Grimmett [EMAIL PROTECTED]:
May 28 01:58:32 WARNING[6821]: chan_sip.c:1830 sip_write: Asked to transmit
frame type 256, while native formats is 4
On Wed, 2005-05-11 at 10:42, Cesar Garcia wrote:
i use asterisk cvs head ( two days ago) more or less
openh323 1.12.2 (oh323 home page)
and
pwlib 1.5.2 (oh323 home page)
asterisk-oh323-0.7.2-pre1
library versions? where download? versions from oh323 readme are not in
sourceforge
I have similar problem on FreeBSD. Gcc and pwlib upgrade solve my
problem...
Regards,
Primoz
-Original Message-
From: Kim Daeyong [mailto:[EMAIL PROTECTED]
Sent: 4. maj 2005 9:11
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] oh323 compile error.
Hi.
I downloaded
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
Hi Michael and Tony
I have the same problem here and I have been able to check that this
problem can be solved disabling VAD in h323 destination routers, I
think this is a common problem with h323 and oh323 modules users and
for me has become a nightmare because my service provider can no
longer
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
The effect that I am seeing is that a call starts
off fine, but suddenly
after a few minutes the audio coming into Asterisk
via OH323 gets very
broken up to the point of being about 90% silence
with occasional brief
snippets of audio getting through.
hi,
any errors or warnings in Asterisk
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:
On Wed, 2005-04-13 at 08:05, 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: warning: passing arg 1 of `ast_channel_unregister'
from incompatible pointer type
Yes, I followed the instructions at:
http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en
Guillermo Salas M wrote:
On Wed, 2005-04-13 at 08:05, Jose R. Ortiz Ubarri wrote:
I have problems compiling the OH323 channel with Asterisk
CVS-HEAD-03/21/05-15:32:10.
I
I just commented that line and everything is ok.
Thanks for help,
JO
Jose R. Ortiz Ubarri wrote:
Yes, I followed the instructions at:
http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en
Guillermo Salas M wrote:
On Wed, 2005-04-13 at 08:05, Jose R. Ortiz Ubarri
Thanks Michael:
I checked but I have the following new compilation errors.
make[1]: Entering directory
`/root/asterisk-oh323-0.7.2-pre1/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE
: Thursday, April 14, 2005 2:49 AM
Subject: Re: [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
I just commented that line and everything is ok.
Thanks for help,
JO
Jose R. Ortiz Ubarri wrote:
Yes, I followed the instructions at:
http://www.oinko.net/astrecipes/index.php?action=artikelcat
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 2:49 AM
Subject: Re: [Asterisk-Users] OH323 and Asterisk
CVS-HEAD-03/21/05-15:32:10
I just commented that line and everything is ok.
Thanks for help,
JO
Jose R. Ortiz Ubarri wrote:
Yes, I followed the instructions
Can you please detail the steps you have taken to successfully compile this
on @home asterisk?
Regards
Mike
- Original Message -
From: CM Rahman Jr. [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, April 09,
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)
Hi,
attached an installation tip from Joao posted the 7th
of jan.
Get oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
Get pwlib from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz
Get
On Thu, 2005-04-07 at 12:54, 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
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] oh323 compilation
On Thu, 2005-04-07 at 12:54, 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
On Thu, Apr 07, 2005 at 12:54:21PM -0500, 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
You can report this here :
https://skylab.inaccessnetworks.com/mantis/main_page.php
Dipole Moment wrote:
Hi all,
I installed and configured OH323 driver and have been using it for a
week now, it's been working great but it also seems to be crashing
Asterisk once in a while. I wasn't sure
Kanishka,
For this question, all your previous questions and
possibly all your future questions, you have to search google
firstor you have to find a consultant, who will help you get
going.
No one can give you a clear and specific answer for a
general question like ' Ido not get the
Dragos Ungureanu schrieb:
...
The redirection itself it's working but nothing is written in the CDR
Hi,
include
amaFlags=billing
and maybe
accountCode=AN_APPROPRIATE_ACCOUNTNAME
into your oh323.conf!
Roger.
___
Asterisk-Users mailing list
Thanks Roger, amaFlags solved my problem
Dragos
On Tue, 2005-02-22 at 12:10, Roger Schreiter wrote:
Dragos Ungureanu schrieb:
...
The redirection itself it's working but nothing is written in the CDR
Hi,
include
amaFlags=billing
and maybe
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
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 like this in the
oh323.conf file or the oh323 documentation.
Hi,
there is a workaround what is doing this job in most cases:
Use as general
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
Then in extensions.conf or better in a file like oh323peers.conf
included in extensions.conf switch to contexts per peer via gotoifs:
This will work ok for my purposes, where I work with CallManager and
gateways.
Thanks,
Adi
___
Asterisk-Users
hi
i used asterisk v1.0.3, pwlib-Janus_patch4 andopenh323-Janus_patch4 with asterisk-oh323-0.6.4 and it worked. it didnt with asterisk-oh323-0.7.0"Rafael J. Risco G.V." [EMAIL PROTECTED] wrote:
HelloI am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4and openh323-Janus_patch4
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
Hi
finally I´ve compiled oh323 0.7.0 with asterisk CVS head version
without any error but now I have troubles trying calls in both
directions:
Xlite(Sip-ua)-Asterisk_oh323--GNUGK(2.0.9)--H323client
Test 1: H323Client call to Xlite:
*CLI -- Executing Dial(OH323/R11652,
Kido,
I start to compile openh323 1.3.15 as recomend in the
asterisk-oh323-7.0 README but I get an error:
h323ep.cxx: In member function `virtual BOOL H323EndPoint::IsLocalAddress(const
PIPSocket::Address) const':
h323ep.cxx:2397: no matching function for call to `PIPSocket::Address::
Thank you very much. I have been trying it but I get in trouble installing it
I will try it again.
Nahuel Ramos.
On Fri, 26 Nov 2004 12:11:20 +0200, Michael Manousos
[EMAIL PROTECTED] wrote:
Thanks.
I appreciate that.
Michael.
kido noagbodji wrote:
i have had some
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
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
Michael Manousos a écrit :
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.
Done
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
: *** [subdirs_build] Erreur 1
What am I missing here..
K.
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, November 25, 2004 11:47 AM
Subject: Re: [Asterisk-Users] oh323 compile issue
administrator tootai a écrit :
[...]
wrapendpoint.cxx:916: no matching function for call to
`H323AudioCodec::IsDescendant (const char *)'
make[1]: *** [wrapendpoint.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.0/wrapper'
make: *** [subdirs_build] Error 1
[EMAIL PROTECTED]
What DTMF mode are you using? Do you have a g729
license installed on
your
system? Remember that g729 only works in passthru
mode.
Can you provide a snip of your extensions.conf file
where you are
trying to
do this?
from K.
If you try to accomplish this with inband DTMFs, then
there is
no
Hi,
But i just can't seem to
make it work using oh323/coded g729? Its like it does
not respond to DTMF signals? I have dig into many
mailing list and not any clear solutions. Could
What DTMF mode are you using? Do you have a g729 license installed on your
system? Remember that g729 only works
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
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:
From what I can tell when I place an outbound call from
Asterisk it always tries to use the first registered H323
alias...
My dial plan in extensions just says Dial(OH323.)
Unlees the gatekeeper rejects multiple calls from Asterisk,
there's no need for multiple
- Non-Commercial Discussion
Subject: RE: [Asterisk-Users] OH323 Trunking
[EMAIL PROTECTED] wrote:
From what I can tell when I place an outbound call from Asterisk it
always tries to use the first registered H323 alias...
My dial plan in extensions just says Dial(OH323.)
Unlees the gatekeeper
Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] OH323 Trunking
[EMAIL PROTECTED] wrote:
From what I can tell when I place an outbound call from Asterisk it
always tries to use the first registered H323 alias...
My dial plan in extensions just says Dial(OH323.)
Unlees
Try, in the 53 (depends on the SW version u're using
voice call send-alert
Also if you're using PRI trunks you can use, in the Serial interface,.
isdn send-alerting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maxim
Litnitsky
Sent: Wednesday,
-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.
Hope someone can
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
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