Displaying 20 results from an estimated 300 matches similar to: "openphone & Asterisk"
2003 Sep 22
2
how to dial a h323 destination ?
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider.
The provider demanded following account data:
H323 ID: XXX-XXX-XX-X
DetinationNumer: XXXXXXXXXXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?
And how i have to make a call with the dial app ?
I have following config:
exten
2005 Jul 27
1
H323 Configuration file
Folks!
I would appreciate if someone could send me a simple working h323
configuration file oh323.conf that is part of asterisk@home
installation.
I have tried to use the oh323.conf content listed on WIKI but it is just
not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot
register. I need a working example of this file for similar phone.
Seshu
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
2004 Apr 13
1
SIP->h323 problem DTMF
I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711).
But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message:
-- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack
--
2004 Sep 04
1
Oh323, Please Help Newbie ;(
Hi,
I just installed OH323 Plugin and im now tryin to make
simple Configuration to connect Openphone and Xlite to
my Asterisk-Server.
All works fine, i just wanna know if there's a
better way to do it? Is there anything wrong with my
Config?
OH323.conf
[general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=8000
udpEnd=8005
fastStart=no
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
2003 Aug 09
2
Gatekeeper
Hello
I am a newbie to Asterisk. We have set up Asterisk on a PC with Redhat 9.0. We have installed H323 openphone on our PC's. We are wondering what a gatekeeper does. It seems we need one but what I have seen in this group is that a gatekeeper must be installed on another box on the network. As all our PC's on the network use Microsoft OS is there a free gatekeeper software for
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 -
2004 Sep 07
0
OH323 return call from openphone to sip?
I figure that I've successfully loaded and compiled the h323 module into
asterisk
I can successfully place a call from openphone to a sip phone (snom200)
So I figure that the h323 module is working.
The question I have is how do I return a call from the sip phone to
openphone?
I get an error message
Sep 7 17:09:49 NOTICE[110992304]: chan_h323.c:861 oh323_request: Asked
to get a
2005 Mar 04
1
Openphone implementation of Speex Codec's descriptions help
Would someone kindly share some definition into the following?
Openphone version 1.91 includes dual sets of Speex codec's starting with:
SpeexNarrow-5.95k{sw}
SpeexNarrow-5.95k{Xiph}
Through
SpeexNarrow-18.2k{sw}
SpeexNarrow-18.2k{Xiph}
I do not understand what the differences are between {sw} & {Xiph} given the
same bit rate for both?
Are all of these Narrow or Wide or Ultrawide
2005 Jan 27
0
Problem with OpenPhone->Asterisk
Hello all,
I just installed Asterisk with H323 support (chan_h323 from Jeremy
McNamara). But experience problem while connecting OpenPhone to Asterisk
Here is h.323 trace:
5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP
Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720,
handle=27
5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225
2003 Jun 24
1
chan_oh323.c Segmentation fault during Openphone/Gnomemeeting connect during module loading...
My apologies if this question has been answered previously. However, I
found that it was nearly impossible to search and find since anything
can cause a segmentation fault.
Problem. When Asterisk is booting up the h323 modules and a client
tries to connect before Asterisk/h323 is finished booting, the program
seg faults out and doesn't load. I thought about putting this into the
inittab,
2007 Oct 05
1
[asterisk-dev] oh323.conf, extentions.conf
Send these questions to Asterisk-Users mailing list.
h323.conf
##################################################
;
; Configuration file of OpenH323 channel driver
;
[general]
listenAddress=W.X.Y.Z ; local ip
listenPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=100
2004 Jul 29
1
OH323 and codec selection
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 terminates the h323 call to an IAX provider
remotely. Asterisk also has G729 licences installed.
in oh323.conf we
2005 May 11
0
Vegastream assistance?
I wonder if anyone can help me?
Am trying to terminate to H323 Vegastream. I'm using OH323 with little
success.
I can dial out and answer but voip end just keepings ringing and ringing.
Thanks for any help.
Neil
Config file:
[general]
listenAddress=ALL
listenPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=no
h245Tunnelling=no
h245inSetup=no
jitterMax=100
2003 Apr 24
1
GnuGK -> Asterisk problem
Hi, i'm trying to setup Asterisk to work with GnuGK using the Openh323
channel driver.
I have a Gatekeeper that gets H.323 calls from a Cisco GW.
To this Gatekeeper I've registered some endpoints, Cisco ATA186, Snom
100, etc.
Now i want send the numbers 083xxx into Asterisk.
Easy, i'll just enter something like this into oh323.conf:
gwprefix=083
And all my calls starting with 083
2003 May 28
0
calls between SIP and H.323 clients
Hello all,
It's me again.
I would like play with calls between a H.323 client and a SIP client
through * inside my LAN.
For that,
on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk;
on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I
register into * with a username, no password. The 3 files oh323.conf,
sip.conf, extensions.conf are in attachment.
In the same
2005 Feb 22
1
how do I dial extensions with oh323?
I have InAccess Networks' oh323 installed and partially working. I can call
the h.323 phone from asterisk using Dial(oh323/${IP_ADDRESS}). How do I
dial from the phone to an asterisk extension? It does not appear to me that
the phone actually registers (or attempts to register) with asterisk.
I'm using Asterisk Stable and the phone in question is a polycom
Soundstation IP 3000 or
2003 May 21
6
chan_oh323.so: Segmentation Fault
Hi,
I'm trying to get H323 support using asterisk 0.4.0
Unfortunately the pwlib and openh323 versions
mentioned in the asterisk-oh323 readme file
are no more available, and I had to use newer
ones.
Now I installed all libraries, but got a
segemntion fault when starting asterisk while
reading the chan_oh323.conf file.
When I declare more than 9 gwprefix I get first
a error "out of
2003 Jun 25
2
no sound pri --> h323
hi all,
i have one (teles) pbx with a BRI telephone and an outgoing E1 port.
The outgoing E1 is connected to an pri_net port from my *.
The incoming call will dail out to a h323 soft phone like openphone or
sjphone or just netmeeting.
The call will be conneted, but i don't hear any sound, from no one of the
both sides.
Can somebody help me?
Thanks,
Thomas.