Displaying 20 results from an estimated 20000 matches similar to: "H.323 (Asterisk@Home)"
2004 Jan 27
0
H 323 + Netmeeting test drive
Hi to everyone,
I am dealing with my primer Asterisk installation and we are trying to
set up a H323 server in order to use Asterisk to place calls
between NM clients (also Gnomemeeting).
I have a basic extensions.conf file:
[general]
static=yes
writeprotect=no
[default]
exten =>
2003 Jul 01
3
H.323 Gateway Connection
Hi,
I'm trying to setup Asterisk to allow users to dial out to the PSTN using a
remote box supporting h.323. I'm using chan_h323.so, and I'm able to make
outbound calls to a client like netmeeting with a line like this:
exten => 242,1,Dial(h323/xxx.xxx.xxx.xxx)
And I'm able to receive incoming calls to asterisk. However I'm not sure how
to route calls to the remote h.323
2005 Aug 06
2
How to test H.323
I'm trying to set-up H.323 support under Asterisk. I built a recent CVS
release and the ooh323c code from the asterisk-addons. Everything built
and installed and the H.323 stuff loads OK when asterisk starts.
What is the easiest way to check if the H.323 code is working? I've
edited the h323.conf and extensions.conf files but I'm sure that things
aren't right. I've
2003 Jun 12
1
Info sip/h.323 interoperability
Hi all,
I'm a student (my thesis work consist in testing
interopearbility SIP/H.323) and I begin to work with
asterisk in this days.
I have to testing to SIP/H.323, since today I have used
Vocal system, but there are some problem for this
features.
In the asterisk mailing list, in the next message I've seen an e-mail
"""
[Asterisk-Users] Cisco
2004 Jun 27
4
H.323 Audio problem UPDATE
Update on this problem:
I gave up on the "native" h.323 because, like others, I couldn't get audio
working. (yes, I tried disabling FastStart in ast_h323.cpp - no change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say
that everything seems to work so far. Not only does audio work, but even
the handshaking is now working in both OpenPhone and even
2004 Aug 13
0
SIP<->H323 "Failed to create smoother"
hello,
Im tryin to make Calls from MS Netmeeting(h323) to
Xlite(SIP) it rings, but as soon as i answered it
dissconnects!!!!
This is what i get from the Asterisk console:
-- Executing Dial("OH323/R27469", "SIP/xlite1|10") in
new stack
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265
create_addr: Setting NAT on RTP to 0
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500
sip_call:
2004 Aug 03
0
OH323 not dial Modem[i4l]/g1
Hello everybody,
I have a strange comportment with oh323 and asterisk, I'start testing
asterisk but with this I can't understant plesae help me !
Thanks
Eltorio
----------------------------------------------------------
1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a
Modem[i4l] line
----------------------------------------------------------
Nothing happens
2005 Jul 07
1
Calls with oh323 with no sound
Hi,
I've oh323 chan installed and working to make calls from SIP to H323
devices. The problem is can no hear sound with the H323 device. I think
this is some related with codecs o nat, because the H323 have one public
IP from a different subnet from the asterisk box.
If I use netmeeting in gateway mode, the call can be completed and I can
talk with a SIP device, but in gateway mode I can not
2006 Apr 04
1
asterisk-ooh323, asterisk 1.2.6 and netmeeting
has anyone managed to get these three beasties to work together ? we're
using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft
netmeeting default from windows xp.
the symptoms are that calls from a SIP client to NetMeeting rings on
NetMeeting, but upon answering the call in NetMeeting, no audio is passed
between the two. eventually, the call times out and hangs up.
on a
2005 Aug 03
0
Asterisk on FreeBSD-5.4 RELEASE : H323 audio problem
Dear All.
I have installed Asterisk-1.0.6 on FreeBSD-5.4 RELEASE via port, so the chan_h323.so modules already included.
When i try the SIP channel, asterisk works fine. In this case, i use the XLite softphone as client, i can hear the voice transfered through clearly. Asterisk is good!! :D
The problem is when i try the H323 channel, the voice cannot be transferred through. I have
2004 Apr 18
0
OpenPhone <-> Asterisk w/H.323
Hello-
In order to satisfy a customer requirement, I've just build H.323 under
asterisk (using the specified versions of OpenH323 & PWLib, and trying to
follow the instructions religiously), and it seems to have come up fine.
When testing with with OpenPhone (Windows version 1.8.1) and NetMeeting,
I've gotten some intermittent results however. All my calls are from a PC
to asterisk -
2005 Jan 06
0
H.323 to SIP extension
Greetings All-
I have an * box with the NuFone H.323 channel driver installed.
I also have an Altigen VoIP system with a PRI to the PSTN.
I can sucessfully make a call from a SIP extension (snom190)
to an H.323 extension (altigen phone)
The thing I can't seem to make work is a call from a H.323 phone
to a SIP extension.
Here's the layout:
2004 Aug 29
0
Asterisk H.323 channel...
Hi all,
I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2).
So far I have been using the H.323 channel included in the tarball (Nufone ?).
I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box :
=====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the
2003 Apr 30
0
H323/GnuGK/Netmeeting
Ok I'm confusEd at this point. I have Asterisk installed with the h323
channel driver compiled and installed. I also installed gnugk. Is it
needed? When I use it as a gatekeeper I'm seeing no interaction with
asterisk. If I dial straight to asterisk with netmeeting I get the
default extension and I can see asterisk trying to play the
"congrats-demo" file but I hear
2004 Sep 20
2
H.323 call problemm (no sound)
Hi all,
I'm having trouble with H.323 outbound calls, * connects but there is no
sound in both ways.
I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which
using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729
licenses installed and this is onli one call.
I tested my * with another ITSP over SIP and G.729 codec and there was
all ok
Here is my configs
2004 Nov 23
1
CLI > h.323 show codecs shows nothing
Hello
I like to make calls to an h.323 device.
I'm using Nuphone h323.
Compiled everything okay "I Guess"
When I make a connection * SIP > h323 device, the phone is ringing and then * tells me "No one available....." and disconnect
Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing.
I try many combination in the h323.conf like.
2007 Feb 04
0
Help sought: Asterisk H.323, Cisco IOS Gatekeeper(s) intra-zone call routing and TETRA
All,
I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...?
I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form
2003 Nov 07
3
Unable dial out with the new Oh323 0.5.6
Hi all,
i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then
i've installed the new chan_oh323 (0.5.6).
when i try to make a call with "netmeeting" through * ( * dial out with
"Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked.
Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7)
installed, and it worked.
Is here
2005 May 25
0
oh323 problems - Solved
For the benefit of everyone, having H323 Configuration problem due to
H245 Tunnel, check the h323 Config embeded at the end. Comment the
offending line as under:
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
-----Original Message-----
From: Tola Ogunsan [mailto:tolaniye@hotmail.com]
Sent: Wednesday, May 25, 2005 1:03 PM
To: Kanuri, Seshu (Company IT)
Subject: RE: oh323 problems
2004 Dec 07
1
H.323 trunking
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:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the