Displaying 20 results from an estimated 1000 matches similar to: "Help with asterisk-oh323 driver"
2005 Sep 21
1
oh323 driver and RFC2833
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 RFC 2833 DTMF relay work over oh323 channel?
Kind regards,
Fernando Herrera
_____
De: Fernando Herrera [mailto:fherrera@iplan.com.ar]
Enviado el:
2005 Aug 29
2
Register Asterisk with Gatekeeper - oh323
I have tried everything. to register with this gatekeeper to make and
receive calls
These are the details I received from the voip provider:
protocol H.323
Gatekeeper Address - AVS@210.21.118.XXX
Port - 1719
RAS - 53
Q931 - 80
h245 - 1722
RTP - 1722
Username - H323
I have 2 phone number/accounts with this gatekeeper that I need to register to.
ID - HMA0200.10szxn-xxxx
e.164 - 22xx2912
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 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
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 22
1
Sip -> H323 using oh323 and G729
Hi All,
I have set up a box that will be used as follows:
SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server
192.168.1.5 192.168.1.50 192.168.1.80
Asterisk is running the latest CVS and oh323 driver.
The SIP phone is a Grandstream Budgetone 100.
I have everything setup and running with G.711 ALAW and ULAW and i'm able
to make calls through
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
2005 Feb 28
2
Asterisk-OH323 no ringing
Hello,
I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5).
Everything is working fine, well, except that : when a call is made from
an h323 device (gnomemeeting for example), the caller does not hear any
ringing at all, he suddenly hears the person who answers the phone.
That can be quite disturbing for the users.
Any help would be very welcome. thank you.
Yves
2005 Mar 09
6
how to sip->h323 using asterisk-oh323-0.7.1
hello
i am using asterisk-oh323-0.7.1. i want to convert sip
call to h323 (h323 sjphone or h323 proxy). what could
be the best way for this. i am successfull in
converting h323->sip by using asterisk as gateway.
help required on sip->h323.
kamran
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2004 Apr 18
1
h323 oh323 g729 please help !
Hello list,
I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. etc
I need: G711 from old phones must be convert to G729 via asterisk and send to provider
I have this problem:
oh323 (last version):
-------------
asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable
2004 Apr 20
1
h323 and oh323 g711 to g729 please help
Hello list,
I have many IP hardphones like Siemens 300 basic ( old ) , cisco
ata.. etc
I need: G711 from old phones must be convert to G729 via asterisk and
send to provider ( G729 from digium )
I have this problems:
oh323 (last version):
-------------
asterisk work with this driver ok for old phones, if I only
faststart=no . But problem with codec , asterisk can speak with
provider (
2005 Mar 20
1
HELP: Failed start after install asterisk_oh323-0.7.1
Hi, ALL:
I install my oh323 channel driver following steps of
http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en
I works my asterisk well before install the chan_oh323.so. But after I
do "make install" the oh_323, my asterisk crash and show me the
following message (asterisk -vvvvvvc).
Does anyone have any idea about it? What's wrong
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323 connections
I am using Asterisk (Debian unstable packages) with an OH323 connection to my
provider. Everything is working except for the generation of ringback tones
when I receive inbound calls from the PSTN. My provider tells me that we're
sending call progress indications and that because of this they're expecting
us to generate the ringback tone. Does anybody know how to configure this in
2004 Nov 26
0
"reason 23 (Temporary failure)" when using Dial(OH323)
I've complied the OH323 .so successfully and can easily receive calls
from my H323 gatekeeper (using 711u), however it seems that all
outgoing calls are refused and I'm getting "reason 23 (Temporary
failure)" as an error code which I can't find documented everywhere.
My H323 gatekeeper needs a 001NXXNXXXXXX to dial out to the PSTN even
if I'm in north america (Montreal)
2005 Aug 18
2
Asterisk (OH323) - gnugk connection
Hello there.
Is there somebody with this connection working? I can't seem to make this
work at all. Could someone
please share some .conf files?
Cheers,
Vedran.
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2005 Sep 30
1
No ringback tone generated by Asterisk with OH323connections
are you giving answer()?
..o-------------------------------------------------------o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose
Comellas
Sent: Friday, September 30, 2005 10:32 AM
To: Asterisk Users
2004 May 04
1
Probs with oh323 driver: audio only in 1 direction
Hi,
try to setup asterisk as an ISDN2H323-Gateway. The only problem
i have after establishing a call is, that Audio works only from IP to
ISDN-Phone but not from ISDN to IP-Phone.
A known problem ???
Thanks in advance
Michael
i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus)
and oh323-0.6.0
Here are my config's
##############
# modem.conf #
##############
2005 Feb 15
0
oh323 question
I'm trying to connect an asterisk server via oh323 to a Lucent iMerge.
I patched the code due so that Lucent can handle asterisk's ver4 h323
http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration
I can now successfully dial in to our company over multiple lines.
The issue is when I dial out
The first outgoing call connects to an outside user A
The second call drops the first
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
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
--