similar to: asterisk+h323+rh9

Hi! I have a runing asterisk box and i want to dial to a analog. pbx using a 4FXS Welltech. Let's say that my pbx have no. 700. If i want to dial to a person in that direction i have to dial pbx prefix (ex. 700), wait for pbx to ansear with hello message and after that to dial internal number(ex. 101). It is possible to dial directly 700101 and asterisk to dial PBX prefix, wait for PBX to

Joao, Thanks for sending the Installation tips as pasted below. It works. Seshu ---------- Get oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz Get pwlib from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz Get asterisk-oh323 from

I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. Firstly, despite the warnings in h323/README, I decided to try using the distro-specific versions of openh323 and pwlib. Of course, the Makefiles in channels and channels/h323 assume that openh323 and pwlib have been specially compiled in $HOME, so I modified the Makefiles to look for headers and libraries in

Hello I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4 and openh323-Janus_patch4 downloaded from inaccessnetworks so I did this: tar -zxvf openh323-Janus_patch4-src-tar.gz cd openh323 patch -p1 < /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch ./configure make opt cd asterisk-oh323-0.7.0 vi Makefile (to set the paths and options according to my system...) NOW I

i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido -------------- next part -------------- An HTML attachment was scrubbed... URL:

Update on this problem: I gave up on the "native" h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even

Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030804/af4203ca/attachment.htm

Hi all, After having tested the SIP part (successful :-)) we are now testing the H323 part of Asterisk. The H323 channel is up and running (using NuFone Network's Open H.323 Channel Driver)) However when dialing to * using ohphone the call can not be set up / established /H323 debug enabled *CLI> == New H.323 Connection created. -- Received SETUP message...

Hello, I am pretty new with Asterisk and I am using it as an H323 gateway.I would like to keep the same h323-conf-id in the outgoing leg as in the incoming leg. So far I have only tried inaccessnetworks' oh323 module, but I think this is a more generic issue. My extensions.conf is pretty simple: [oh323_context] exten => _XXXXXXXXXX,1,Dial(oh323/${EXTEN},30,tr) So my question is the

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

Hi R-Help, I would like to make filled contour maps of ocean data overlaid by costlines from the map package. I can draw the filled contours and the coastlines om the same plot, but the filled contour also covers part of the land. To get rid of that I tried to draw a filled coastline map on top of the filled contour, but the filled map only draws the closed contours - so most of the land is

Hello, I just downloaded and installed the latest version of asterisk under Fedora. (had it under FreeBSD but was having TOOO many problems) After my installation i noticed that the channel H323 was not included ( I remember that i did not have to install it under freeBSD) but I have seen that SIP and IAX are supported though. So i am wondering: Does asterisk consider H323 so achaic that it does

Hin-Tak Leung wrote: > Peter Dalgaard wrote: >> brechbuehler at gmail.com wrote: >>> Full_Name: Christian Brechbuehler >>> Version: 2.4.1, 2.5.1, OS: Ubuntu GNU/Linux >>> Submission from: (NULL) (24.61.47.236) >>> >>> > <snipped> >>> Example (start R without DISPLAY from bash): >>> % DISPLAY=3D R >>> >

I m new to asterisk n i've got an IP phone that supports h323 protocol.... but i dont know how to configure asterisk to use it... i m comfortable in using sip & iax softphones.... but there is no h323.conf in /etc/asterisk/ .... i read that i've to compile some files but i m confused regarding h323 & oh323 ...... which one should i use.. plz tell me or atleast give some helpful

Hello, I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 installed but failed. I applied the patch to the required OpenH323 library according to the instructions, and set the proper directories in the Makefile. Here is what I receive after I issue make: ******************************* g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -

Hi I want to run a Samba PDC with LDAP backend on a FreeBSD 6.2 for Sparc64. And off course, if I send this mail, it is not working ! :-( I have this error message when using smbclient on the PDC itself. The command line is: smbclient -L janus -Uadministrator%toto The result is: session setup failed: Call returned zero bytes (EOF) I've tested on 2 differents FreeBSD6.2 Sparc64, I have the

Hi Jean-Marc, We already decided to use Marvell's 88F8618 platform, But I have few questions to you and listed it below, Q1, Can we implement the Speex Codec on Marvell 8618 for non-HW DSP of platform? Q2, Based on ARM9 CPU core, do you have any information to let me estimation the voice computing power for Speex codec? Sorry to bother your job again.... Best Regards Janus ( Freedom )

Thanks a lot. Well, the problem is that it's hard to find any informaton about converting *.mjr files. I'll keep trying. Regards, Mateusz 2015-07-09 0:57 GMT+02:00 Mark Harris <mark.hsj at gmail.com>: > Hi Mateusz, > > It looks like the Janus code that writes Ogg Opus will not produce > correct Ogg Opus output when there are packets missing. You can quote > me or

Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing "asterisk -vvvc". Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on