I have a problem when dialing from outside line to sip server. I get this
output on debug.
Could someone give me a hint what could be wrong?
== Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to
exten 's'
== Starting OH323/R30149 at from-pstn,s,1 still failed so falling
Title: Message
Hi
all,
Of course I am a newbie, so please bear with
me...
I'm having a lot of trouble getting things to work
properly and I am sure it is a configuration issue somewhere, I'm just not sure
where...I have been all through my extensions.conf and cannot seem to see a
Title: Message
Hi
all,
Of course I am a newbie, so please bear with
me...
I'm having a lot of trouble getting things to work
properly and I am sure it is a configuration issue somewhere, I'm just not sure
where...I have been all through my extensions.conf and cannot seem to see a
You need to post your extensions.conf and oh323.conf for further assistance.
It sounds like though that the h.323 endpoints are sending a call to
you and since you didn't define a default extension/context for them
to go to, they are trying to go to extension 's' in the default
context, but
Title: Message
Hi
all,
All outbound calls
work perfect from my SIP ATA
186...
SIP ATA 186 (w/
private IP) Asterisk H323 (public IP) TDM works
perfect
TDM H323
(public IP) Asterisk SIP ATA 186 (w/ private IP) fast busy with
this error message in the CLI of Asterisk:
May 17
Hello all
i'm build success H322 in channel/H323 of asterisk. but don't know how to use it.
i run GNUGK on server and client using ohphone. when i dial to asterisk server. the connection accept and disconnect.
please help me to configure in H323.conf and extensions.conf.
Ok, at the bottom of my h323.conf file on my 1st server I have this:
; -
[test]
type=user
host=209.237.227.185
context=termination-test
incominglimit=10
accountcode=005
; -
Using an Asterisk at the other IP, I have this:
exten =
hello
i successfully installed asterisk on fedora core 3 and all what's in the
check list plus the ACTOS gui and asterisk manager but i used actos to
configure my cisco ip phones and dial/receive calls through sip.
my problem is i need to configure H323/Fax in asterisk to catch H323/Fax
from
my problem is i need to configure H323/Fax in asterisk to catch H323/Fax
from the gateway and route it as t38/fax to another pbx server i
installed on windows.
how can i configure, route and convert the faxes?
___
Asterisk-Users mailing list
Anyone using H323 on asterisk on a larger
scale. For example a few million minutes a month? I would like to
hear about your experience good or bad.
Thanks,
Jon
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Yes. I'm using Asterisk CVS-v1-0-01/24/05 with asterisk-oh323-0.6.5
compiled in on two GW's with two DS1's doing RBS wink in each box
which then flip the calls to G711ulaw/H.323 back to an Avaya S8700
where the users are at.
Using the Varion quad span cards, and the APIC/IO-APIC kernel
enabled,
Of BJ Weschke
Sent: Friday, April 15, 2005 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk-Users@lists.digium.com
Subject: Re: [Asterisk-Users] H323 Large Scale
Yes. I'm using Asterisk CVS-v1-0-01/24/05 with asterisk-oh323-0.6.5
compiled in on two GW's with two DS1
I have a connect to * via H.323/g711 from device A and want to connect
to B which want for H.323/g729
h323.conf contains
disallow=all
allow=alaw
allow=g729
but outgoing faststart/TCS contains only g711 (from h323_request(format)
i think) and so no codec negotiation and no voice.
Howto run up
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Orehov Pasha wrote:
I have a connect to * via H.323/g711 from device A and want to connect
to B which want for H.323/g729
h323.conf contains
disallow=all
allow=alaw
allow=g729
but outgoing faststart/TCS contains only g711 (from
: George K. Konstantoulakis [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 23, 2005 3:11 AM
Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk
compertable
Hello Bashir,
what kind
Hello Bashir,
what kind of problems are you having with oh323 ?
George
Bashir Ullah - www.Lamsre.Com wrote:
Hi All * lover.
This is not a question only this is a request to all SIP and Asterisk user .
I am also with asterisk last few month and providing callingcard soluation.
most of the SIP or
: Tuesday, March 22, 2005 11:57 PM
Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk
compertable
If you use open-source software, you have to accept that sometimes
project need some times to be stable and have all features.
OH323 works - even if there are still a few bugs
PROTECTED]
- Original Message -
From: George K. Konstantoulakis [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 23, 2005 3:11 AM
Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk
: George K. Konstantoulakis [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 23, 2005 3:11 AM
Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk
compertable
Hello Bashir,
what kind of problems are you
Hi all,
I'm new to asterisk and had just install it on my linux
server.
Can anybody told me how to setup it up for interworking with
cisco h323 voip gateway?
I check throught the manual on http://www.digium.com/downloads/marketing/asterisk.pdf
but cannot find any information for
hi!
u can find more info here www.voip-info.org
/madhawa
On Tue, 22 Mar 2005 16:51:20 +0800, raymond [EMAIL PROTECTED] wrote:
Hi all,
I'm new to asterisk and had just install it on my linux server.
Can anybody told me how to setup it up for interworking with cisco h323 voip
Hi All * lover.
This is not a question only this is a request to all SIP and Asterisk user .
I am also with asterisk last few month and providing callingcard soluation.
most of the SIP or IAX provider asking very high price which is really tough
to resell in market. but still there is some h323
If you use open-source software, you have to accept that sometimes
project need some times to be stable and have all features.
OH323 works - even if there are still a few bugs - and the people around
the project are working hard to make to work even better.
If you want something that work now,
Hi all,
I am wondering if chan_oh323 or chan_h323 supports NAT traversal the
following setup:
H323 phone - Asterisk --- NAT router - H323 gateway - PSTN
I am trying to register a H323 gateway through a NAT to Asterisk for
outgoing calls to PSTN.
How can I achieve the above?
This is possible. But success depends also on whether the router can do port
forwarding and whether the H323 Gateway supports NAT.
This is possible with Quintum for instance with some port forwarding rules on
router level.
Selon VoIP Newbie [EMAIL PROTECTED]:
Hi all,
I am wondering if
Thanks. Is there any native solution that is also cheap? I need it for
my small office with only a few staff. My H323 gateway is not even a
cisco one but costs only $200.
Thanks.
On Mon, 21 Mar 2005 15:57:13 +0100, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
This is possible. But success
Hello,
I'm trying to register with a gatekeeper using chan_h323, I have a
Login, Password and a user (telephone) number,
This is my h323.conf
[general]
port=1720
bindaddr=x.x.x.x (my fixed IP)
gatekeeper=x.x.x.x (gk ip)
allowgkrouted=yes
allow=all
[xx] (my login)
type=h323
e164=111 (my
hi guys im getting this error when trying to load chan_h323 on my local
box
Mar 16 17:19:27 WARNING[2278]: libh323_linux_x86_r.so.1.12.2: cannot
open shared object file: No such file or directory
Mar 16 17:18:36 WARNING[2265]: Loading module chan_h323.so failed!
any ideas? everything compiled
hi folks,
I am calling from Asterisk to some Cisco gateways (as5350, 2600) and I am
having one way problems with chan_oh323.
With other provider that uses cisco also and I running the same
chan_oh323, it works perfectly.
I've tried also with chan_h323 and it does not work as well.
Asterisk cvs
Good day all
Can asterisk connect h323 clients to each other and h323 to sip and what
about h323 video?
Please Help and advice
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Altus,
Yes, Asterisk can do the following scenarios, amongst others:
Client -- H.323 -- Asterisk -- H.323 -- Client
Client -- H.323 -- Asterisk -- SIP -- Client
In these scenarios, it is acting as a Back To Back User Agent (BTBUA).
It can also handle video calls, though I have not used this
On Tue, 2005-02-15 at 13:59 +, Alistair Cunningham wrote:
It can also handle video calls, though I have not used this myself.
AFAIK video only with SIP, which I didn't test myself either. With
H323 it does not work, audio only there.
Regards, Bruno.
Hi all,
How can I configured H323 EPs or OH323 EPs to get them authenticated
through GNUGK???
Many thanks
Ben
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Could you help me with this problem? When I call H323 gateway there is no
sound in both ways.
Here is h323 debug:
- begin
-- Executing Dial(SIP/msn-6297, H323/[EMAIL PROTECTED]:1720) in new
stack
Allowed Codecs:
Table:
G.729A{sw} 1
G.729{sw} 2
Good day all
I have a asterisk server running sip and sip phone
How do I get asterisk to call another h323 server?
Please Help
Thanks
Altus
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Good day all
Just to re phrase my previous question
We have asterisk running sip for sip phone
In the US there is a h323 server
What I want to do is:
All calls coming into my pbx via sip thats got a american number to go
threw the h323 server
I have set this up with 2 sip servers where the one
I got My ASTCC kind of working, but the problem I have is
that it tries to send all the calls over SIP.
How can I configure it with H323?
Thanks
KF
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
You can do that in your extensions.conf for the context
you are using for astcc originated calls
Seshu Kanuri
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Krystian
FiliksSent: Thursday, January 20, 2005 5:43 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users
Someone have had good luck compilig h323 into YDL?
first thinked was a bug in code but twisted said it is
wierd - isn't that the recursive pthread lib? If so, do you have the
kernel development headers/libs installed?
I've instaled kernel source, what more can i do? any help would be very
PROTECTED] On Behalf Of Walid Azab
Sent: Sunday, January 16, 2005 8:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] H323 Softphone for iPAQ
Hi list,
I was just wondering, is there any H.323 soft-phone that can be
installed on a pocket PC (iPAQ).
Walid
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] H323 Softphone for iPAQ
Also the following has worked great for me:
http://www.wifive.net/introduction.asp
Michael
Radovan Mihalik wrote:
http://www.sjlabs.com/sjp.html
SJphoneR is a VOIP softphone that allows
Hi
list,
I was just
wondering, is there any H.323 soft-phone that can be installed on a pocket PC
(iPAQ).
Walid
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
-
Non-Commercial Discussion'
Subject: [Asterisk-Users] H323
Softphone for iPAQ
Hi list,
I was just wondering, is there any
H.323 soft-phone that can be installed on a pocket PC (iPAQ).
Walid
___
Asterisk-Users
Guys, I am
about to install H323 [EMAIL PROTECTED] (Asterisk
CVS-v1-0-12/22/04-05:48:41). I noticed that the default h323h.conf file is not
set up. I also noticed that many of you here say that it is better to use
Oh323.
What is the best
scenario here for me?
Should I go with the
Walid Azab wrote:
Guys, I am about to install H323 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
(Asterisk CVS-v1-0-12/22/04-05:48:41). I noticed that the default
h323h.conf file is not set up. I also noticed that many of you here say
that it is better to use Oh323.
What is the best scenario
: [Asterisk-Users] H323 on [EMAIL PROTECTED]
Walid Azab wrote:
Guys, I am about to install H323 [EMAIL PROTECTED] mailto:[EMAIL
PROTECTED]
(Asterisk CVS-v1-0-12/22/04-05:48:41). I noticed that the default
h323h.conf file is not set up. I also noticed that many of you here
say that it is better
Do the paths to each of the include files exist?
If not, you will need to edit the Makefile in that directory to point
to the right include directories.
- James
On 18/12/2004, at 1:14 PM, David Adade wrote:
Hi,
Can anyone help? I get the following error when trying to complie the
h323
On Mon, 20 Dec 2004, James wrote:
Do the paths to each of the include files exist?
If not, you will need to edit the Makefile in that directory to point
to the right include directories.
- James
On 18/12/2004, at 1:14 PM, David Adade wrote:
Hi,
Can anyone help? I get the
Hi,
Can anyone help? I get the following error when trying to complie the h323
channel under the source installation directory
asterisk/channels/h323
i have read the readme file and kept to the recomended versions; h.323 v1.12.2
and PWLIB v1.5.2
Thanks in advance
[EMAIL PROTECTED]
Francisco wrote:
Hi, im getting mad compiling the H323 channel (Jeremy's version
inAccess version). Ive tryed many versions of openh323 lib and pwlib,
and i get differets errors.
Does anyone uses this channel? and which version of it, openh323 lib and
pwlib?
asterisk-oh323-0.6.4 compiles/works
Hi, im getting mad compiling the H323 channel
(Jeremy's version inAccess version). Ive
tryed many versions of openh323 lib and pwlib, and i get differets
errors.
Does anyone uses this channel? and which version of
it, openh323 lib and pwlib?
Currently im using Linux Slackware 10.0, and i
I am attempting to set my outgoing calls through H323, however once the call
is picked up, no audio works on either side.
Any suggestions?
H323.conf
--
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=g729
allow=gsm
dtmfmode=rfc2833
gatekeeper = 63.208.156.185
We are in urgent need of some help getting Asterisk to gateway between an
incoming H323 connection and SIP to a Lucent TNT. We have the incoming
H323 already set up and the SIP going to the TNT but the media stream is
getting lost somewhere as no audio is heard. We are willing to pay $$$ for
an
Peter,
Peter Landy wrote:
New to Asterisk so I am sure this has been answered before. I can
compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323
then I get all kinds of errors even though I have set the paths up in
the source files. I can attach the errors if it is useful. I
Peter,
If you have the lastest CVS version of asterisk(1.0.11) , and the latest
version of asterisk-oh323(0.7.0), it won't work.
What version of asterisk are you running? what version of oh323 are you
trying to compile?
K.
Peter,
Peter Landy wrote:
New to Asterisk so I am sure this has
perfectly it is just H323 I am hung up on.
Cheers
Pete
-Original Message-
From: Michael Manousos [mailto:[EMAIL PROTECTED]
Sent: 22 November 2004 09:01
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] H323 Problems
Peter,
Peter
-Commercial
Discussion
Subject: Re: [Asterisk-Users] H323 Problems
Peter,
Peter Landy wrote:
New to Asterisk so I am sure this has been answered before. I can
compile PWLIB and OpenH323 but when it comes to compiling
asterisk-oh323 then I get all kinds of errors even though I have set
the paths
Peter Landy wrote:
Yes I did. Does anyone have a working list of libraries and versions. I have
tried with different releases of H323 and they all give different errors.
Also is it necessary to compile the H323 under asterisk src/channels/H323
as this also bails on errors. The rest of my asterisk
Hi!
i have to make pabx to direct calls to h323 terminals. i have an h323
gateway available and wish to use asterisk as the gatekeeper for call
direction and queueing etc.I am a beginner at asterisk and to link
openh323 with asterisk for my project i searched on net i found
different compilation
New to Asterisk so I
am sure this has been answered before. I can compile PWLIB and OpenH323 but when
it comes to compiling asterisk-oh323 then I get all kinds of errors even though
I have set the paths up in the source files. I can attach the errors if it is
useful. I though however that
,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Landy
Sent: Sunday, November 21, 2004 11:01 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] H323
I spent today trying to get openh323 working with Asterisk 1.0 on my new
AMD64 box. I ran into a number of problems. The first being that the
openh323 build scripts do not recognize x86_64 as an architecture that it
builds on. I hacked the scripts appropriately and got it built. I set the
Hi,
We have a setup of the Nufone Implementation of H323 on our Asterisk Setup and
it appears to work fine apart from one slight technical glitch.
When a customer makes a call, it keeps trying to forward the client into the
default context inspite of a context=blah in the particular config.
Any
Hi Folks,
I have two H323 Polycom video conference system with a Linux firewall
Iptables in the middle. I am not getting to make H323 working in this
setup and I was wondering to put two * servers as a bridge to jump
the firewall using IAX.
The idea basically is:
h323 Polycom
Hi,
-Original Message-
I have two H323 Polycom video conference system with a Linux
firewall Iptables in the middle. I am not getting to make
H323 working in this setup and I was wondering to put two *
servers as a bridge to jump
the firewall using IAX.
The idea basically is:
Asterisk and IAX to bridge it. Asterisk wont pass Video
in its H323 only audio.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isamar
Maia
Sent: Sunday, November 14, 2004 8:21 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] H323/*/IAX - Firewall
Setting up a Gatekeeper can be a pain. After looking at Speed Dial /
New Context from Wed, 3 Nov 2004 18:24:31, I added the following bits
into 'extensions.conf'.
Maybe useful to others..
In my incoming default profile - I have...
; Calls from the H323 Extentions
exten =
Hi list,
I face 2 problems with a today (11/04/04) CVS version:
H323:
Nufone h323: calling Dial(H323/extension) give in logs
Host: extension Username:
placing outgoing call to :1721 - this is my GK port
h323_make_call failed(H323/extension)
[...]
dialstatus=chanunavail
Calling from a
I'm assuming nobody has experience with running ISDN / BRI over H.323...
-Original Message-
From: Huddleston, Robert [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 03, 2004 8:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] H323 ISDN
Hi List!
I am able to buy and test a wireless basestation that can connect
standard digital cordless phones to a voip system.
The basestation is available in 2 models each supporting a
different protocol to speak to the PBX (ether H323 or SCCP).
Any advice what the best protocol would be, H323
Hi,
-Original Message-
I am able to buy and test a wireless basestation that can connect
standard digital cordless phones to a voip system.
The basestation is available in 2 models each supporting a
different protocol to speak to the PBX (ether H323 or SCCP).
Any advice what
On Mon, 1 Nov 2004, Florian Overkamp wrote:
Hi,
-Original Message-
I am able to buy and test a wireless basestation that can connect
standard digital cordless phones to a voip system.
The basestation is available in 2 models each supporting a
different protocol to speak to the PBX (ether
Hi,
On Mon, 2004-11-01 at 18:29, Remco Barende wrote:
I won't be (able to) doing any debugging, but I could try several
protocols. I checked the wiki and there are several issues for both
protocols. The oh323 protocol is tested with several types/brands of
equipment, on the SCCP onlt a
-Original Message-
I am able to buy and test a wireless basestation that can connect
standard digital cordless phones to a voip system.
The basestation is available in 2 models each supporting a
different protocol to speak to the PBX (ether H323 or SCCP).
Does any one know
I know Innovaphone - http://www.innovaphone.de - has nice (but not cheap)
H.323 DECT gateways that support multi-cell roaming.
l.
In data Mon, 01 Nov 2004 11:20:06 -0800, TC [EMAIL PROTECTED] ha scritto:
-Original Message-
I am able to buy and test a wireless basestation that can
Subject: Re: [Asterisk-Users] H323.conf question
sometimes its like pulling teeth
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 11:00 AM
Subject: RE: [Asterisk-Users] H323
... Can't authorize the calls based on IP :-(
I'm just quickly answering. Not sure if it's the same problem.
On July 28 th, i was no more able to dial ip iax clients.
It was working before. Since, I need to dial them with their
account number. I told Mark, but he told me that it's a iaxcomm
Hello
I have chan_h323.so compiled.. And got is up and running
I can place calls now from my cisco AS5350 to asterisk and back
Only..
In h323.conf it doesn't seem to 'see' the my user .. It's just always
using the default context
If I dial from 192.168.1.50 (my cisco) to 192.168.1.100 (my
i would just lose the line where it says context=default323
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 5:52 AM
Subject: [Asterisk-Users] H323.conf question
Hello
PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, October 26, 2004 1:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] H323.conf question
i would just lose the line where it says context=default323
Thanks,
Steve Totaro
[EMAIL PROTECTED
Subject: RE: [Asterisk-Users] H323.conf question
Ofcourse I tried that :-)
In this case an h323 call to asterisk doesn't work anymore (at all)..
The asterisk debug window is then complaining that it can't find a
default context at the moment I set up a h323 call
Thx
Niels
-Original Message
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, October 26, 2004 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] H323.conf question
Work around is to create a default323 context and use a goto
Maybe I dont understand the problem fully. What does the console show?
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 8:12 AM
Subject: RE: [Asterisk-Users] H323
Subject: Re: [Asterisk-Users] H323.conf question
Maybe I dont understand the problem fully. What does the console show?
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 8
output from console?
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 9:09 AM
Subject: RE: [Asterisk-Users] H323.conf question
I want to route to the right context
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] H323.conf question
output from console?
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 9:09 AM
Subject: RE
sometimes its like pulling teeth
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 11:00 AM
Subject: RE: [Asterisk-Users] H323.conf question
When reloading: it just
: [Asterisk-Users] H323.conf question
sometimes its like pulling teeth
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 11:00 AM
Subject: RE: [Asterisk-Users] H323.conf
Hi Everyone
We would like to connect our Splicecom Maximiser PBX to our Asterisk box
via H323 so that we can send our US calls via a low cost carrier (e.g.
Broadvoice).
Has anyone managed to do this in the past (I remember seeing some
companies also worked with this system in the UK).
The
Hello Jeremy,
thanks for your remark..
this is what i get out of it ...
--
#0 0x41f57c50 in oh323_new (i=0x80f8f50, state=0,
host=0x449e5147 213.xxx.202.xxx) at chan_h323.c:625
625 chan_h323.c: No such file or directory.
in chan_h323.c
#0 0x41f57c50 in oh323_new
Hi there !
I searched the whole web to find some helping information about H323
Control Protocol, but there is no way to find that information.
We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2
+ 'asterisk-oh323_1.5 channel driver + wrapper' and configured the
dialplan for
Hello Asterisk list;
when i DIAL(H323/[EMAIL PROTECTED]) i get this strange error
--
-- Executing Dial(SIP/home-0953, H323/[EMAIL PROTECTED]|5|r) new stack
backupns*CLI
Disconnected from Asterisk server
--
Asterisk just goes down..
--
Best regards,
Danny
Danny Zak wrote:
Hello Asterisk list;
when i DIAL(H323/[EMAIL PROTECTED]) i get this strange error
--
-- Executing Dial(SIP/home-0953, H323/[EMAIL PROTECTED]|5|r) new stack
backupns*CLI
Disconnected from Asterisk server
--
Asterisk just goes down..
You are going to have to provide more debug
HI there !
Need help, I'm using asterisk 0.9.0,pwlib 1.5.2,openh323_1.12.2 and
asterisk_oh323_1.5. All H323 Endpoint can dial each other for 30 sec,
after that connection lost because of an H323 Control Protocol Error !!!
this is the asterisk output while phoneing :
Hi there !
I searched the whole web to find some helping information about H323
Control Protocol, but there is no way to find that information.
We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2
+ 'asterisk-oh323_1.5 channel driver + wrapper' and configured the
dialplan for
Hi there !
I searched the whole web to find some helping information about H323
Control Protocol, but there is no way to find that information.
We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2
+ 'asterisk-oh323_1.5 channel driver + wrapper' and configured the
dialplan for
[EMAIL PROTECTED] wrote:
Hi there !
I searched the whole web to find some helping information about H323
Control Protocol, but there is no way to find that information.
We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2
+ 'asterisk-oh323_1.5 channel driver + wrapper' and
Hi,
is there a way to force a user authentication using h323 channel from
asterisk sources? Do I have to use gatekeeper for this?
Is there any way to do it in h323.conf just like in sip.conf?
eg:
[mazek]
secret=xx
auth=md5
tia
mazek
--
http://www.marcinmazurek.com/ ::: nic-hdl:
Hi,
did anybody managed to compile h323 channel under Fedora 2? There's only
gcc 3.3 and 3.4. Does h323 from * or opencall work with FC2 and gcc 3.3?
Anybody had similiar problems?
tia
mazek
--
http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE
GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4
Does asterisk support using an H.323 provider for outgoing calls? From
everything I have found, it looks like it does. However, I have had no
success in getting it to work. I would really appreciate if somebody
could give me a hand. I am using the channel that comes with asterisk.
I have
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