Dear all;
I have an incoming call from Ericsson PBX to Asterisk through H323 trunk. I
need to transfer this call back to Ericsson and then
Asterisk should release the channel so that if I shutdown Asterisk call should
not be disconnected. As far as I know Transfer function does not work over
Dear all;
I have an incoming call from Ericsson PBX to Asterisk through H323 trunk. I
need to transfer this call back to Ericsson and then
Asterisk should release the channel so that if I shutdown Asterisk call should
not be disconnected. As far as I know Transfer function does not work over
oh yes, i'm using h323 not openh323
On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote:
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote:
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr
try
UserByAlias=yes in general and type=user in user context.
On Fri, Apr 26, 2013 at 9:48 AM, s m sam.gh1...@gmail.com wrote:
oh yes, i'm using h323 not openh323
On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote:
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote:
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from two
side. but it is not good for me because 200 is the name of extension and
when i config asterisk
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in
Hi list,
I've been beating my head for about 3 days on this one. I have
Asterisk 1.4.41 installed using openh323. As long as I'm inside my
firewall, everything is hunky-dory. When I move to server on another
subnet, I'm still able to connect, but no longer have sound. Any good
pointers
Danny Nicholas wrote:
Hi list,
I've been beating my head for about 3 days on this one. I have
Asterisk 1.4.41 installed using openh323. As long as I'm inside my
firewall, everything is hunky-dory. When I move to server on another
subnet, I'm still able to connect, but no longer have
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jose P. Espinal
Sent: Wednesday, April 27, 2011 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] h323
[Danny Nicholas]
Thanks for the information - but this doesn't seem to play well with SUSE.
Any ideas?
If you are open to the possibility of building from source I think I
might have a little white paper based on the scripts (about installing
latest version of H323plus on 1.4.X) by today,
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jose P. Espinal
Sent: Wednesday, April 27, 2011 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] h323
On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi shinacaly...@gmail.comwrote:
Hi!
I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide
a conference bridge for an existing Avaya PBX. I have no control over the
Avaya system, but I am able to speak with the admin in charge
I have posted my problem on the link below, but didn't get any answer. I am
hoping someone here can help me with this issue. Here's my problem:
I am using H323 to talk between Asterisk and Avaya IP Office 500. For
some strange reason, when we are talking on a VoIP call, we get
disconnected
Nobody on this ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: September-16-09 7:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] H323 RTP Transmission error of packet
Using H323 to reach
Using H323 to reach another h323 switch, I have no audio and the following
error:
[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transmission error of packet 21282 to XXX.XXX.XXX.XXX:6064: Invalid argument
[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Hi,
Still I can manage to have good incoming calls from h323. Can someone give
me a hand?
Regards,
LS
Date: Thu, 16 Jul 2009 15:46:43 +0100
From: Luis Silva luis.si...@dreamware.pt
Subject: [asterisk-users] H323 situation
To: asterisk-users@lists.digium.com
Hi all,
I have this installation:
Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0.
I have a problem that is, when a call comes from H323 and goes to a Sip
phone the asterisk sends two rtp streams to the sip. I checked this with
tcpdump, save the payload (voice is in
Maybe this can help you? http://astrecipes.net/index.php?n=286
Thanks
l.
2009/5/31 Tamer Higazi th9...@googlemail.com
Hi people!
I am looking for a h.323 implementation guide for asterisk. I looked in
the doc folder of the latest asterisk source distribution and I didn't
fund anything
Hi people!
I am looking for a h.323 implementation guide for asterisk. I looked in
the doc folder of the latest asterisk source distribution and I didn't
fund anything acording to this subject.
If you guys could give me any advise, I would thank you.
Tamer
Hello, Im using channel_h323 by Jeremy McNamara to connect my asterisk box
to an Gatekeeper and I want to do some filter by remote ip addres but I
dont know what variable in asterisk have this data. Someone knows how is
the name or which are the name of this variable in channel h323? Thanks for
Hello,
We made small stress-test for H323.
Test shows that H323 protocol is heavyweight compared with SIP.
More details: http://wiki.kolmisoft.com/index.php/H323_pass-through_test
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Hello,
Asterisk 1.4.18.1
PWlib 1.10.0
Openh323 1.18.0
../asterisk/channels/h323 compiled from source.
Under high load H323 crashes and kills Asterisk, debug shows:
(gdb) bt
#0 0x007a2b18 in strcmp () from /lib/libc.so.6
#1 0x014478a1 in find_call_locked (call_reference=13,
Dear all,
Does asterisk supports H323?If yes how to enable it?
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Yes, this has already been answered. Search previous post for
implementation.
On Thu, Oct 9, 2008 at 3:34 AM, michel freiha [EMAIL PROTECTED] wrote:
Dear all,
Does asterisk supports H323?If yes how to enable it?
Regards
___
-- Bandwidth and
hi.
i have two IP phones that are in H323 protocol. How can i test that
these two phones are working? For IP phone (SIP) i used asterisk
server. can i use asterisk server to test the ip phone with H323
protocol.
--
Mahboob Zaman
System Engr
Systems Services Limited
Cell: +8801712280308
Yes you can.
Obviously you have to compile, configure and add chan_h323 to Asterisk.
Map
On Thu, Aug 28, 2008 at 10:32 AM, mahboob zaman [EMAIL PROTECTED]wrote:
hi.
i have two IP phones that are in H323 protocol. How can i test that
these two phones are working? For IP phone (SIP) i used
Hi,
Thanks for reply. can u give me information in detail? How can i compile and
can i add chan_h323 ?
Thanks
mahboob
On 8/28/08, map [EMAIL PROTECTED] wrote:
Yes you can.
Obviously you have to compile, configure and add chan_h323 to Asterisk.
Map
On Thu, Aug 28, 2008 at 10:32 AM,
Of mahboob zaman
Sent: 28 August 2008 12:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] H323 protocol
Hi,
Thanks for reply. can u give me information in detail? How can i compile and
can i add chan_h323 ?
Thanks
mahboob
On 8/28/08, map
El jue, 28-08-2008 a las 01:32 -0700, mahboob zaman escribió:
hi.
i have two IP phones that are in H323 protocol. How can i test that
these two phones are working? For IP phone (SIP) i used asterisk
server. can i use asterisk server to test the ip phone with H323
protocol.
I've wrote a
Hey,
I'm not sure whats going on but I have built and installed chan_ooh323
from asterisk addons. When I try to dial a call to an h323 provider i
get the Channel not implemented error.
When I load chan_ooh323.so I get:
[Aug 10 14:28:00] WARNING[23007]: loader.c:647 load_resource: Module
On Sunday 10 August 2008 13:31:22 emist wrote:
I'm not sure whats going on but I have built and installed chan_ooh323
from asterisk addons. When I try to dial a call to an h323 provider i
get the Channel not implemented error.
When I load chan_ooh323.so I get:
[Aug 10 14:28:00]
Thanks Tilghman, that was the issue.
Regards,
Igor H.
Tilghman Lesher wrote:
On Sunday 10 August 2008 13:31:22 emist wrote:
I'm not sure whats going on but I have built and installed chan_ooh323
from asterisk addons. When I try to dial a call to an h323 provider i
get the Channel not
I have following settings done on my Fedora8:
Downloaded
openh323-v1_19_0_1-src-tar.gz
pwlib-v1_11_1-src.tar.gz
Extracted them in /root/openh323 and /root/pwlib
Exported the following variables:
PWLIBDIR=/root/pwlib
export PWLIBDIR
OPENH323DIR=/root/openh323
export OPENH323DIR
Hi,
I have following settings done on my Fedora8:
Downloaded
openh323-v1_19_0_1-src-tar.gz
pwlib-v1_11_1-src.tar.gz
to my knowledfe chan_h323 should be compiled against
openh323-v1_18_0-src.tar.gz
and
pwlib-v1_10_3-src-tar.gz
cheers
--
I am after someone to help me to config H323 on asterisk if possible since I
am far too busy stuck on another project. Interested parties please msn me
on sam _ _ tam AT hotmail.com please take out all space and change AT to @
If you are unsure then you can always email me with your contact via
Hi List;
In the h323.conf file, the parameter gatekeeper is
used to let asterisk work as h323 gatekeeper listening
at port 1719 by setting gatekeeper=DISCOVER or it is
used to let asterisk search for the gatekeeper to talk
with it and receive calls from it? But if just to let
asterisk talk with
I have not tested it but in theory you should be able to authorize it by
setting host= in the peer details.
- Original Message -
From: bilal ghayyad [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 09, 2007 11:14 PM
Subject: [asterisk-users] H323
Hi All;
As I understood that h323 module in asterisk does not
support the ability to let the h323 endpoints register
at asterisk (this registeration happens at 1719 port),
so how asterisk will be able to route the call for the
destination IP Phone if it is not registered (so the
IP is unknown)?
We've configured ooh323 on our 1.4.6 asterisk server.
We've looked at various sites for tips, most recently
http://www.tek-tips.com/viewthread.cfm?qid=1243330page=3. The module
seems to load properly. When we do a tcpdump, we see traffic flowing
between the asterisk server and the Avaya
Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 07, 2007 8:40 PM
Subject: [asterisk-users] h323 problem with asterisk 1.2.18
i am experiencing problem with asterisk 1.2.18
I've downloaded and installed pwlib and openh323 with the following
commands:
cd /path/to/pwlib
Hi,
I am using Asterisk 1.2.9.1, with chan_h323.
The problem I am coming across is when trying to bridge an incoming
H323 call with another H323 call:
phone1 dials into asterisk with H323, for extension 111
in asterisk:
exten = 111, 1, Dial(chan_h323, H323/[EMAIL PROTECTED])(in my
i am experiencing problem with asterisk 1.2.18
I've downloaded and installed pwlib and openh323 with the following commands:
cd /path/to/pwlib
./configure
make clean opt
cd /path/to/openh323
./configure
make clean opt
then 'ive set the corresponding PATH
Did you compile H.323 for asterisk and then make install asterisk ?
- Original Message -
From: Pezhman Lali [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 28, 2007 4:30 PM
Subject: [asterisk-users] h323
hi
After compiling and installing pwlib
hi
After compiling and installing pwlib and openh323 ,
the asterisk, give the folloing error.
please tell me where the problem is ?
Best
Mani
*CLI -- Executing Dial(SIP/2.2.2.2-086f5ac0,
H323/[EMAIL PROTECTED]|60) in new stack
Mar 28 14:17:23 WARNING[11985]: channel.c:2576
ast_request: No
Hi all,
I have some questions about h323. Is it mandatory to install a oh323 or I can
do h323 calls without patching or adding any new drivers ti asterisk?
I did compile the asterisk with channel driver chan_h323 but it still gives me
this error:
[Feb 28 18:12:58] WARNING[1902]: app_dial.c:1081
Florea Igor wrote:
Hi all,
I have some questions about h323. Is it mandatory to install a oh323 or I can
do h323 calls without patching or adding any new drivers ti asterisk?
I did compile the asterisk with channel driver chan_h323 but it still gives me
this error:
[Feb 28 18:12:58]
I need to receive a FAX call from a SIP device into my Asterisk box, then send
that FAX call to an H323 gateway and bridge the call, so Asterisk will be
acting as a Converter.
SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the
H323 gateway only supports T.38
BTW, i
: [asterisk-users] H323-to-SIP proxy
I need to receive a FAX call from a SIP device into my Asterisk box, then
send that FAX call to an H323 gateway and bridge the call, so Asterisk will
be acting as a Converter.
SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but
the H323 gateway
@lists.digium.com
Subject: [asterisk-users] H323-to-SIP proxy
I need to receive a FAX call from a SIP device into my Asterisk box, then
send that FAX call to an H323 gateway and bridge the call, so Asterisk will
be acting as a Converter.
SIP device is a Grandstream HT496 so i can configure FAX Pass
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] H323-to-SIP proxy
What about the SIP leg?
- Mensaje Original -
De: Michelle Dupuis [EMAIL PROTECTED]
Para: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviados: martes 27 de febrero de
I'm looking at setting up an asterisk box dedicated to SIP-H323 conversion
(a 3rd party is currently converting the protocols for us).
1. Is it worthwhile to split this functionality onto a second server? Or
should we let the ast pbx handle the conversion? (we have a couple hundred
active
Which H.323 channel driver are you using, and could you post a log or debug
of a session.
Craig
- Original Message -
From: Andrei U [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 08, 2007 2:41 AM
Subject: [asterisk-users] H323 to SIP - One way voice
Hello all,
I want to use asterisk as protocol converter, H323 to SIP. I am using
Asterisk 1.2.14 with chan_h323 and the free version of g729.
When calling from SIP to H323 everything is fine. But when calling from H323
to SIP, the phone using SIP doesn't hear the other party.
The phones and
Hello all,
I want to use asterisk as protocol converter, H323 to SIP. I am using
Asterisk 1.2.14 with chan_h323 and the free version of g729.
When calling from SIP to H323 everything is fine. But when calling from H323
to SIP, the phone using SIP doesn't hear the other party.
The phones and
I thinik the code is too new for your compiler... I remember reading about
needing GCC 2.95 somewhere... I'm just about to post on a similar theme!
I am getting the following compile error on h323.
Its an old redhat 7.3 system with asterisk 1.2.14, zaptel 1.2.12
pwlib 1.5.2 and openh323
I am getting the following compile error on h323.
Its an old redhat 7.3 system with asterisk 1.2.14, zaptel 1.2.12
pwlib 1.5.2 and openh323 1.12.2
I have pwlib compiled and installed.
I have openh323 compiled and installed.
I went in the channels/h323 directory and did make opt
What shall I
Hi,
I installed asterisk with oh323.
My gatekeeper is out of nat device.
How can i register * to gatekeeper?
Thanks in advance..
Jason.
Cheap talk?
Check out Yahoo! Messenger's low PC-to-Phone call rates.
I dont think the registration will be the problem, but the media
communication, for that you could use an Application Layer Gateway
(ALG), you can check netfilter.org for more information.
Regards
On 12/1/06, Jason Kim [EMAIL PROTECTED] wrote:
Hi,
I installed asterisk with oh323.
My
Hi,
My configuration is SipPhone-asterisk1
-asterisk2.
My asterisk version is 1.2.10.
I installed chan_h323 according to
'http://astrecipes.net/?n=102'.
When i call from asterisk1 to asterisk2, there is no
audio.
Using 'rtp debug', I can see that rtp packets are
being received.
Regards,
Hi
The communcation between an alcatel telephone switchbox and a sip phone (using
asterisk h.323 implementation) isnt working fully bidirectional.
The user at the alcatel telephone switchbox can hear the user who is speaking
on the sip phone but not the other way around.
Could that be a
Hi guys!Can someone give advice on nice H323 IP phones brands?? I'm looking for some H323 IP phones for a customer. Diving in theinternet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard about them. Can someone give feedback experince about it??, configease, sound quality, visual
On Sunday 27 August 2006 10:40, Mohammad Salaque wrote:
any one try that with g723 codec?
We use G.723.1, and it works well. My only problem is the
bridging time (after pickup) takes at least 5 seconds.
But this happenned even before Asterisk was in the picture, so
I'm guessing it's the
any one try that with g723 codec?
thanks
Salaque
On 8/27/06, Rosli Sukri [EMAIL PROTECTED] wrote:
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem
to get it to work with ms netmeeting
On 8/26/06, atik khan [EMAIL PROTECTED] wrote:
Hi,
i used to work ooh323 with
Hi
What is the best solution for H323 in asterisk
-- h323 in source,
-- oh323 or
-- ooh323c?
which is most robust and reliable? Which supports gatekeeper functionality?
Best wishes
Andrutto
--
Najnowsze fakty!!!
Hi,
i used to work ooh323 with my asterisk. it gives better performance
than other oh323 or H323 comes with asterisk...
i got H323 channel and oh323 with a lot of error.( like codec
selection )but ooh323 works fine with me
thanks
atik
On 26 Aug 2006 12:13:52 +0200, andrutto
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeetingOn 8/26/06, atik khan
[EMAIL PROTECTED] wrote:Hi,i used to work ooh323 with my asterisk. it gives better performance
than otheroh323 or H323 comes with asterisk...i got H323 channel and
Hi,all:
in /etc/asterisk/h323.conf
I setting
gatekeeper=192.168.0.19
secret=3001
and on server 192.168.0.19 I running a openh323gk
and add a user
3001 and password is 3001 too, but when I booting asterisk, I
got
messages :
Error registering with gatekeeper
I have a requirement to set up an Asterisk server that will
handle H323. In the end this is used for video conferencing but it will be
transitioning other H323 devices to SIP at some point. My question is this:
Does anyone know of or have good documentation that explains how this
I recently have been required to terminate traffic via H323. We have beensuccessfully handling this traffic as SIP. We often have 30 + concurrent calls on this server and I am not quite sure the best way to handle this
new H322 traffic. Which of the h323 channels for * can handle
Joshua Laroff wrote:
I recently have been required to terminate traffic via H323. We have
beensuccessfully handling this traffic as SIP. We often have 30 +
concurrent calls on this server and I am not quite sure the best way to
handle this new H322 traffic. Which of the h323 channels for *
I'm trying to setup an Asterisk box as an H323 to SIP gateway.
Basically, I'd like to receive traffic in H323 and forward to another
Asterisk box (on the same network) using either IAX2 or SIP so that
the second Asterisk box communicates with other gateways using SIP.
Therefore, if I
I installed an asterisk server with oh323 channel driver support.
Then I uploaded the H323 firmware on a AT320 phone (Usually I use it as a
sip phone, but I am using it just for test)
Let's say that I assigned 945 as phone number, account and password to this
phone, and its ip address were
Everyone,
I have been trying to connect a PBX with H323 IP trunks
with g711 codec to my Asterisk server running ooh323 service. I can place calls
to and from either the Asterisk, or PBX with no problem, but when I try to
pickup the call on either end, the phone hangs up immediately.
Finally I installed the oh323 without any errors and tested that with SJPhone.(Played the demo message).
Now my question is, it seems from any h323 client anyone can make calls
to my asterisk if they dial number@my serverip.
How do I do the authentication by IP, username, password like SIP.conf
Hello all,
I am trying to use native h323 built from asterisk 1.2.7. I configured
the h323 to receive incoming calls...the problem is i can receive the
call to my asterisk and it rings another extension but no audio. I
don't see any good documentation about gatekeepers, fast start, etc
with h323.
Hello All,
Somereason my previous mail was not get into the list (or may be
delayed). I have a problem successfully configuring the h323 support
with asterisk 1.2.7.
I searched the net and I don't find any useful or clear documentation.
First tell me, which h323 installation should I go with?
Hello all,
I am trying to use native h323 built from asterisk 1.2.7. I configured the
h323 to receive incoming calls...the problem is i can receive the call to
my
asterisk and it rings another extension but no audio. I don't see any good
documentation about gatekeepers, fast start, etc
Hi yousuf,
Please tell me to make h323 work, what are the other things i need to do other than getting the chan_h323.so under modules?
Do I need to install OpenGatekeeper and configure it ?
Do I need fast start? fast tunneling? h245inSetup? (I really don't have any idea about what these components
If I were you, I would install the lastest asterisk-addons. there is an asterisk ooh323c directory , read the REAME on that directoryThameem Ansari [EMAIL PROTECTED] wrote: Hello All, Somereason my previous mail was not get into the list (or may be delayed). I have a problem successfully
It seems that Open H323 only work with Asterisk version 1.0. As per the
latest stable README of asterisk-oh323 here is the readme.
Required packages
---
In order to build the OH323 Asterisk channel driver you will need
some other packages. We recommend to download their source
On Thu, Jun 08, 2006 at 04:58:20PM -0700, Thameem Ansari wrote:
It seems that Open H323 only work with Asterisk version 1.0. As per the
latest stable README of asterisk-oh323 here is the readme.
Which h323? chan_oh323 is just one of at least three h323 channels.
Versions 0.7x of it are for
And addons package includes chan_ooh323c . Unlike the latter two it does
not use openh323 and thus a lot simpler to build (assuming you have
gcc-objc).
gcc-objc? IIRC, the Objective System OOH323 is written in plain C(99?)
not Objective-C. If they wrote it in Objective-C, they would be
Hello guys,
Thanks for your replies. I finally got the ooh323 built successfully.
But again the problem is I am using sjphone to connect to my server. I
can initiate the call which rings the phone without any problem. But
its keep on ringing even if I take the call. I dunno whats goin on?
Simply
On Thu, Jun 08, 2006 at 08:23:15PM -0700, Thameem Ansari wrote:
Hello guys,
Thanks for your replies. I finally got the ooh323 built successfully. But
again the problem is I am using sjphone to connect to my server. I can
initiate the call which rings the phone without any problem. But its keep
On Saturday 20 May 2006 16:31, Roman Yeryomin wrote::
Hello all!
I have a problem with ringing indication when calling from h323 (oh323+open
phone client) to sip users. The phone rings and users can talk to each
other with no problems but the calling h323 user hear silence unless sip
user
Hello all!
I have a problem with ringing indication when calling from h323 (oh323+open
phone client) to sip users. The phone rings and users can talk to each other
with no problems but the calling h323 user hear silence unless sip user picks
up the phone.
Calling to pstn no problems. Calling
Daren J. Howell DTCommunication wrote:
I have restricted the asterisk server to G711 to match the choice on the
PBX, and still same result.
I have read that either endpoint have to be either a master or slave to
communicate to each other. I see in the logs that both are shown to be a
Daren J. Howell DTCommunication wrote:
Have Asterisk connected to a H323 compatible legacy PBX using QSIG
protocol and IP trunks.
I can call to Asterisk, and from Asterisk using X-Lite softphone but
whenever either end picks up, the calls disconnects.
Try restricting both ends to one
I have restricted the asterisk server to G711 to match the
choice on the PBX, and still same result.
I have read that either endpoint have to be either a master
or slave to communicate to each other. I see in the logs that both are shown to
be a slave. The pbx side has to be set to slave.
Have Asterisk connected to a H323 compatible legacy PBX
using QSIG protocol and IP trunks.
I can call to Asterisk, and from Asterisk using X-Lite
softphone but whenever either end picks up, the calls disconnects.
No gatekeeper is installed. I have attached a copy of my
h323 logfile for
Farhad Ibragimov wrote:
I don’t have practice to work with Asterisk but I see that is a great soft.
If you have any idea or some config files can you help me
Asterisk is perfectly documented everywhere on the net. Maybe the first
place to visit in order to have working asterisk is
Hi all
I have installed station which support only H323
protocol. I want to install SIP telephone. Is it possible to call SIP telephone
throught my station
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Asterisk-Users
You could make a H323 to SIP transport. Before to do that, you need to
have installed and working both chan protocolos on Asterisk.
aFarhad Ibragimov escribió:
Hi all
I have installed station which support only H323 protocol. I want to
install SIP telephone. Is it possible to call SIP
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] H323 to SIP
You could make a H323 to SIP transport. Before to do that, you need to
have installed and working both chan protocolos on Asterisk.
aFarhad Ibragimov escribió:
Hi all
I have installed station which support only
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