On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote:
Hi list!
I asked this in this list some time ago, and now I was searching for
evolution about this subject, but I found nothing.
Nowadays, what is the state for H.323 video support?
Is there support in the 1.6 beta
Yes, you are right... sorry for my fast and poor English.
I rewrite my questions:
Nowadays, what is the state for H.323 video support?
Is there support in the 1.6 beta branch?
If not, is this in the roadmap for 1.6 branch?
Regards.
2008/5/23 Steve Totaro [EMAIL PROTECTED]:
On Fri, May 23,
Remind me to pick on your poor Spanish next time I see you for a
mid-morning meal. :)
Steve Totaro wrote:
On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote:
When and where is the 1.6 brunch? ;-)
___
-- Bandwidth and
Hi,
I have used h323, oh323 and ooh323.
My experience is that ooh323 does not work properly, i dont recommend it.
I dont know why, but the sound is bad, with sound breaks. I also need
to put some wait (2) functions after the answer( ) or playback( )
functions, it think that asterisk takes some
Hi,
I'm using H323 in asterisk 1.4.9
work well
On 8/3/07, yonoko molomo [EMAIL PROTECTED] wrote:
Hi,
I have used h323, oh323 and ooh323.
My experience is that ooh323 does not work properly, i dont recommend it.
I dont know why, but the sound is bad, with sound breaks. I also need
to put
Hi there,
I have use the H.323 module that comes with asterisk-addons and i
consider it (so far) VERY stable for my needs.
Im talking about 10,000 minutes at month , + or - , and never had a
crash or something bad about it.
Personally, i recommend it,
--
J. P.
rakh at slackware-es dot
You need install the asterisk h323 drivers. You can get them in the
asterisk-addons.
- Original Message -
From: bilal ghayyad [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, June 22, 2007 9:49 PM
Subject: [asterisk-users] H.323 IP Phones and H.323 Traffic
Hi List;
On 24 Jul 2006, at 14:24, Asif Ali wrote:
Hi
I have a problem with the NAT using H.323 and am thinking of
employing IAX as a workwround. I have a scenario in my mind which I
am not sure is gonna work or not, neaways here it goes.
I want my IAX clients to connect to Asterisk which will be
I had problems with sjphone ... same version as yours.
Finally, i managed to solve it by:
- in sjphone, media channels settings: untick Use remote codec
preferences and Open audio streams after remote opened ... it was
trial-error ... now it works (to Echo and Sip-H323 call).
- in asterisk,
Hallo Cesc
Cesc writes:
I had problems with sjphone ... same version as yours.
Finally, i managed to solve it by:
- in sjphone, media channels settings: untick Use remote codec
preferences and Open audio streams after remote opened ... it was
trial-error ... now it works (to Echo and
Hi,
I'm trying to connect * to Nortel BCM 50, This PBX use H.323
v3 to interface with other PBX. The port use to connect is TCP 1720 but
I can't configure this port on my * box. I'm using a H.323.conf file
sample to activate the port but the * isn't listening there. Somebody
Cenk,
Are you sure that remote will handle H245
tunneling? If the remote does not know how to do that, you will get transport
failure. I would suggest doing FastStart instead and
see if you are getting the same results. Of course, you can verify that the
remote can handle faststart as
Cenk Yabas wrote:
Thanks to Yves's commitment I was able to configure oh323 channel, cleared
the codec issue, registered to Gatekeeper, placed a call, but receive this
message on the console. What might be the problem?
Asterisk Ready.
*CLI -- Registered with gatekeeper '[EMAIL PROTECTED]'.
On Tue, 2005-02-01 at 14:09 +0900, Kuniyoshi Murata wrote:
Hi,
I'm thinking of setting up Asterisk for H.323 video phone clients.
Now, what is the difference between native H.323 that come with Asterisk and
Open H.323 for Asterisk ?
I can't tell you the exact differences, but oh323
Sebastian Nocetti wrote:
is h323 per user based working??? I have setup this:
[User1]
type=user
host=xx.xx.xx.xx
context=international
incominglimit=30
But all calls from xx.xx.xx.xx are not routed to context
international, it is working?
I am using chan_h323
I'm using current
Thanks !! I will try!!
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Soren Rathje
Enviado el: Martes, 21 de Diciembre de 2004 02:30 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] h.323 Type=User
Sebastian
Now it is working... Thanks!
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Soren Rathje
Enviado el: Martes, 21 de Diciembre de 2004 02:30 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] h.323 Type=User
See below.
Nardis Dome wrote:
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:
--- Michael Manousos [EMAIL PROTECTED]
wrote:
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323
trunk.
I am trying to set up the following scenario:
hi michael,
thx for the answer, but now i have the following
error:
Executing Dial(SIP/2004-b1cf,
OH323/192.168.204.130) in new stack
-- H.323 call to 192.168.204.130 with codec ALAW
-- Called 192.168.204.130
-- H.323 call 'ip$localhost/11490' cleared, reason
24 (Call ended with
Jeremy McNamara wrote:
Mészáros Mihály wrote:
Please if you can please help me to solve this problem.
Help yourself and READ THE README.
Hello Jeremy!
I read it already! ;-) thx!
But i didn't find a word about that chan-h323 what decoder encoder use.
It use the libopenh323 or other in built
Mészáros Mihály wrote:
Please if you can please help me to solve this problem.
Help yourself and READ THE README.
Jeremy McNamara
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
[EMAIL PROTECTED] wrote:
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
: Senad Jordanovic [mailto:[EMAIL PROTECTED]
Sent: Monday, September 20, 2004 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] H.323 call problemm (no sound)
[EMAIL PROTECTED] wrote:
Hi all,
I'm having trouble with H.323 outbound calls, * connects
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] H.323 call problemm (no sound)
[EMAIL PROTECTED] wrote:
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
Jeremy McNamara wrote:
Michael Manousos wrote:
The performance of the oh323 channel driver is limited by OpenH323.
asterisk-oh323 uses the (more complete) RTP implementation offered by
the library, and not that of Asterisk. Of course there are pros
(adaptive jitter buffer, RTCP implementation)
Scott Stingel a écrit :
Hi-
In answer to your questions:
Someone on Friday had said that disabling Fast Start corrected the audio
problem with H.323, so yesterday I tried to disable it in
~/asterisk/channels/h323/ast_h323.cpp. Today, I noticed that Jeremy
(NuFone) uploaded a new version of this
Tommy,
Still waiting from you whether the CDRs are recorded with cdr_csv.
This is working just fine for me.
Michael.
T. Chan wrote:
Hi, Scott. Are you telling me that this native h.323 has been hardcoded
with fast start? Can you tell me where in ast_h323.cpp it is that you
disabled this faststart?
Sorry this has nothing to do with your audio issue, but I noticed you were
able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with asterisk-oh323
0.6.2. I get the following errors when trying to compile the oh323 wrapper
for asterisk:
-- snippet of errors --
In file included from
Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Wilkins
Sent: Monday, June 28, 2004 8:15 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H.323 Audio problem UPDATE
Sorry this has
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you were
able to
Michael:
Yes I did.
On Yaum al-Ithnain 10 Jumaada al-Awal 1425 11:28 am, Michael Manousos wrote:
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Ok,
I got it all to work finally. I removed everything and started from
scratch. I also got the latest version of asterisk from the CVS. I built
PWLib, then applied the patch to oh323 1.13.5 then built oh323, and finally
built and installed the wrapper (0.6.3). I just started up Asterisk
Michael Manousos wrote:
The performance of the oh323 channel driver is limited by OpenH323.
asterisk-oh323 uses the (more complete) RTP implementation offered by
the library, and not that of Asterisk. Of course there are pros
(adaptive jitter buffer, RTCP implementation) and cons (lower
Hi, Scott. Are you telling me that this native h.323 has been hardcoded
with fast start? Can you tell me where in ast_h323.cpp it is that you
disabled this faststart? Have you tried using the Stable cvs of the
Asterisk.
Can you let me know which version of the OH323 are you using ? Is it the
: [Asterisk-Users] H.323 Audio problem UPDATE
Hi, Scott. Are you telling me that this native h.323 has been hardcoded
with fast start? Can you tell me where in ast_h323.cpp it is that you
disabled this faststart? Have you tried using the Stable cvs of the
Asterisk.
Can you let me know which version
Scott Stingel wrote:
Some load testing to following this week, but I'm encouraged!
This is where you are going to be discouraged with that other H.323
driver. I guarantee it.
Disabling fast-start has solved the problems for quite a few other ppl
using 5300s, so you must be doing something
stably could be different, please share, thanks.
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Sunday, June 27, 2004 9:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H.323 Audio problem UPDATE
Scott Stingel wrote:
Some
Just checking that you have installed the proper versions of both OpenH323
and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt
asterisk after those installations as specified?
If so, then you are having the same problem I'm experiencing: no audio on
H.323. I'm also
FYI
I am experiencing the same problem.
I have complied asterisk from the latest CVS
The call connects with no audio or DTMF to either end.
I tested with ulaw and g729 with no success.
-Michael
On Fri, 2004-06-25 at 10:55, Scott Stingel wrote:
Just checking that you have installed the
I have the same problem here. I have to servers working with identical
(same) configurations, the old one is working just perfect and the new one
I got, is not working (no voice in both directions). Im trying to fix this
problem with digium, we are exchanging emails so if I get a solution Im
gona
This is a known architectural issue that does not appear to have been
resolved yet.
See bug http://bugs.digium.com/bug_view_page.php?bug_id=0001337
The problem is the mapping of the various internal states of different
Asterisk channels on to the Q.931 states.
Asterisk currently does not wait
Ive placed a bounty on my bug. See http://bugs.digium.com/bug_view_page.php?bug_id=0001334
From: Derek Samford
Sent: Wednesday, April 07, 2004
4:26 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] H.323
Seg faulting
Can someone take a look, tell me if this is a bug,
See the existing discussion on this
Ditto.
IT DOES NOT WORK. Compiles, but no calls go through. I asked you to post
your exact versions of all components, but I don't believe you did this. I
have not been able to get it to work with Asterisk 0.7.2. Just because
*YOU* got it to work on your
On Saturday 06 December 2003 19:01, Greg Boehnlein wrote:
Hello,
I have a friend that is asking if he can use his Ericsson 3413
H.323 IP phone with Asterisk. I can't seem to find any reference to this
phone on the Wiki...
you can either use chan_h323 or chan_oh323 (the latter is contrib
Roy Sigurd Karlsbakk wrote:
I know people are running h.323 in production (or so I've heard), but as (AFAIK) there still are some
unsolved issues, YMMW.
Why not list out the specific problems and they can be addressed if they
are still a problem? You are quick to bash my channel driver but
On Sat, 6 Dec 2003, Jeremy McNamara wrote:
Roy Sigurd Karlsbakk wrote:
I know people are running h.323 in production (or so I've heard), but as (AFAIK)
there still are some
unsolved issues, YMMW.
Why not list out the specific problems and they can be addressed if they
are
THERE IS AN INCONSISTENCY IN THE README FILE THAT IS OUT OF DATE:
Follow the instructions on line below and do NOT issue a make clean
install in asterisk/channels/h323 as indicated elsewhere, just issue a
make and then in /usr/src/asterisk (or wherever you source is), issue
a make install and
Look again, this time with the cvs code.
Jeremy McNamara
Paul Cheng wrote:
THERE IS AN INCONSISTENCY IN THE README FILE THAT IS OUT OF DATE:
Follow the instructions on line below and do NOT issue a make clean
install in asterisk/channels/h323 as indicated elsewhere, just issue
a make and
Dear all,
I just fresh CVS the asterisk code, and uncomment the G729 in the Makefile
on asterisk/channels/h323.
I also donwload pwlib and openh323 from nufone.net/downloads, and did
following things:
1. /pwlib, make clean, make both
2. /openh323, make clean, make opt
3.
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of G Lin
Sent: Thursday, November 06, 2003 8:33 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] H.323 and G729: Another sad tale
Dear all,
I just fresh CVS the asterisk code, and uncomment the G729 in the Makefile
G Lin wrote:
Dear all,
I just fresh CVS the asterisk code, and uncomment the G729 in the Makefile
on asterisk/channels/h323.
I also donwload pwlib and openh323 from nufone.net/downloads, and did
following things:
1. /pwlib, make clean, make both
make opt
Anything else just wastes massive
Jeremy McNamara said:
2. /openh323, make clean, make opt
3. /asteriks/channels/h323, make clean, make install, and it is got
error
about no chan_h323.o exists. and the make install is failed.
You haven't read the README
And I quote:
To compile this code:
Issue a make in the
I can also confirm chan_h323 and g.729 work well to 5300s, but we had
to build on RH8 not RH9. Haven't tried 5300 to Asterisk
except via SIP
which works fine--even to i4l interfaces.
I believe that when you use up2date on both RH8 and RH9, you end up with the
same version of Kernel. So
I can also confirm chan_h323 and g.729 work well to 5300s, but we had
to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP
which works fine--even to i4l interfaces.
On Friday, October 31, 2003, at 01:57 AM, Jeremy McNamara wrote:
John Todd wrote:
I've done some reviewing of
John Todd wrote:
I've done some reviewing of the archives for G729 and H323
experiences. The landscape of that query isn't pretty - lots of pleas
for help, and nor do I see too many answers. I have a pending bid
that requires some data before I can implement * on this particular
solution.
Quoting Jeremy McNamara [EMAIL PROTECTED]:
Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be
terminated on Asterisk systems and sent out Zap interfaces?
IMHO as for today No,
For incomig I couldnt even get it working with g711 and ciscos 72xx and as5300.
Calls were dropped from
Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be
terminated on Asterisk systems and sent out Zap interfaces?
IMHO as for today No,
For incomig I couldnt even get it working with g711 and ciscos 72xx and
as5300.
Calls were dropped from cisco side after two udp packets from
Hello,
Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be
terminated on Asterisk systems and sent out Zap interfaces?
A while ago, I only manage to get g729 call works when terminating in Cisco
AS5300 from Asterisk but was unable to terminate call in Asterisk from Cisco
AS53000
Olaf Menzel wrote:
But I want to transmit the original callerid as defined in sip.conf
via the H.323 gatekeeper to a H.322 phone. How to manage this ??
How about Dial,H323/[EMAIL PROTECTED]/${CALLERIDNUM} ?
However you will have issues with the gatekeeper if it is expecting a
specific H.323
.
- Original Message -
From: Roy Sigurd Karlsbakk [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 6:22 PM
Subject: Re: [Asterisk-Users] H.323 - SIP gateway
You shouldn't treat asterisk as a gatekeeper (because it ain't) On your
H.323 equipment, set asterisk
I am trying to configure * to route calls from SIP extension to an
externeal H.323 gatekeeper and vice versa.
The route from * to the gatekeeper is a simple ENUM call and work fine:
[outbound][outbound]
exten = _3XXX,1,Dial,H323/[EMAIL PROTECTED]
One Snom100 phone is defined in sip.conf:
[snom]
1719, try to change h323.conf:
[general]
port = 1719
bindaddr = 0.0.0.0
Regards,
Gus
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 12:23 PM
Subject: Re: [Asterisk-Users] H.323 - SIP gateway
h323 runs
PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 12:23 PM
Subject: Re: [Asterisk-Users] H.323 - SIP gateway
h323 runs on port 1720. Your gatekeeper is trying to contact the
wrong
port number.
On Tue, 2003-10-14 at 10:02, Mireia Munoz de jesus wrote:
Hi all
]
[mailto:[EMAIL PROTECTED] Behalf Of Bryan Nolen
Sent: 03 October 2003 14:55
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] H.323-SIP Gateway
Basically you just need to make sure that the (o)h323 channel is compiled.
Personally I use the chan_oh323 driver (google it).
Its very easy, just
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 22 September 2003 04:02, Jeremy McNamara wrote:
You have to enable ring indications
exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr
That doesn't work when you use H323 directly. As in
Dial(H323/ip$12.34.56.78|120|r) ... Works fine
I have found that mixing the Dial() format with | can cause problems.
Does Dial(H323/ip$12.34.56.78,120,r) work as expected?
On Mon, 2003-09-22 at 03:09, Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 22 September 2003 04:02, Jeremy McNamara wrote:
You have
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 22 September 2003 10:16, Eric Wieling wrote:
I have found that mixing the Dial() format with | can cause problems.
Does Dial(H323/ip$12.34.56.78,120,r) work as expected?
Doesn't change anything.
Here's a better explanation of the
You have to enable ring indications
exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr
Jeremy McNamara
Roy Sigurd Karlsbakk wrote:
hi
seems like things are closing in to something that might look like
success. I have one problem left: I don't get ring indicator when I dial
out from the
chan_h323 is built into asterisk. Check the /usr/src/asterisk/channels/h323
directory for more info.
- Original Message -
From: Phillip Britt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 02, 2003 1:12 PM
Subject: [Asterisk-Users] H.323 Support
Hi,
I am currently
Jeremy == Jeremy McNamara [EMAIL PROTECTED]:
Jeremy What part of IN OTHER WORDS: Run Open H.323 v1.11.7, nothing
Jeremy newer, nothing older if u want this to work. don't you
Jeremy understand?
Well, I was trying to find out (politely) about some things. Please
allow me to paste from my
What part of IN OTHER WORDS: Run Open H.323 v1.11.7, nothing newer, nothing older if
u want
this to work. don't you understand?
Jeremy McNamara
Jan Rychter wrote:
I have hit a problem where chan_h323 sometimes doesn't hang up properly
and stays stuck in the Up state, with asterisk consuming
Hi Justin,
Try:
exten=242,1,Dial(h323/[EMAIL PROTECTED])
Regards,
Szymon Czyz
Justin Eckhouse [EMAIL PROTECTED] wrote:
Hi,
I'm trying to setup Asterisk to allow users to dial out to the PSTN using a
remote box supporting h.323. I'm using chan_h323.so, and I'm able to make
outbound
exten = _91XX,1,Dial(H323/${EXTEN:[EMAIL PROTECTED])
${EXTEN:1} will grab all the digits you sent in 91XX and the :1, in
${EXTEN:1}, tells it to drop the first digit.
Michael
I'm trying to setup Asterisk to allow users to dial out to the PSTN using a
remote box supporting
Justin Eckhouse wrote:
exten = 244,1,Dial(h323/xxx.xxx.xxx.xxx/PSTN-NUMBER-HERE)
This is bad... if you use this kind of exten line PSTN-NUMBER-HERE will
be the H.323ID Asterisk will use to make the call.
exten = 244,1,Dial(h323/[EMAIL PROTECTED]) is the proper
format.
Jeremy McNamara
Julio Tommasi wrote:
Have any body succesfully compiled the files in
asterisk-oh323-0.2.tar.gz ?
This is a very, very old version.
Try the latest one (0.5.1) from
http://www.inaccessnetworks.com/projects/asterisk-oh323
Michael.
I have the following errors:
+for x in wrapper asterisk-driver; do
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