similar to: First demo between IAX2 and chan_h323 works !

Hi all, I try to ask helps again about chan_h323 extensions. I define this in h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 allow=gsm allow=ulaw gatekeeper = DISABLE context=default [gm1] type=friend host=192.168.1.20 context=default [gm2] type=friend host=192.168.1.25 context=default and I have in extensions.conf : [demo]

Hello all, Encouraged by the successful "demo", I'am getting on with Asterisk CVS. I added 2 H.323 extensions in extensions.conf [default] include => demo exten => 701,1,Dial(H323/gm2@192.168.1.20/s) exten => 702,1,Dial(H323/gm2@192.168.1.25/s) With: - [demo] is defined by default in sample.extensions.conf - Asterisk server is running on host 192.168.1.20, on the

Hello all, I'am trying to use 2 gnophones on my LAN. But I can't get any sound. Here is my configuration: 1. on PC1, I have: - Asterisk compiled from CVS (CVS-05/15/03) and it runs. - Gnophone binary version from Debian: 0.2.4+cvs.20020624-3 - a sound card working ( module es1371 for /dev/dsp) - I registered it as "alice" in extensions.conf 2. on PC2, I

Hello all, I tried to compile Asterisk from CVS yesterday. I would like to try gnophone as IAX client on the same Asterisk server with a sound card. I didn't have any problem of compilation. Here are the config files in /etc/asterisk: /etc/asterisk/asterisk.conf /etc/asterisk/extensions.conf /etc/asterisk/iax.conf /etc/asterisk/indications.conf /etc/asterisk/logger.conf

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

Hello! Tried to compile, but : [CC] chan_h323.c -> chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing pointer to incomplete type chan_h323.c:350: error: dereferencing pointer to incomplete type chan_h323.c: In function 'oh323_rtp_read': chan_h323.c:790: error: dereferencing pointer to incomplete type chan_h323.c:791: error:

Hello all, I got the following error compiling h323 support in the latest cvs. Below the error is a diff to the file that I got to make it work. I took an example out of sip as far as the syntax for ast_rtp_new. Not sure if it is correct or not, but it seems to work. Please correct me if I am wrong in the additional 2 arguements. Regards, Scott cc -g -pg -c -o chan_h323.o -march=i686

Hi All, I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris 10. When I compile asterisk, the build fails at chan_h323 with: ---------------------------------------------------------------------- chan_h323.c: In function `reload_config': chan_h323.c:2863: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_h323.c:2863: error: (Each undeclared

Hello List, I've been trying to compile Asterisk with H.323 support and, after correctly installing PTLib and H323plus (OpenH323), the Asterisk configure script still doesn't detect the dependencies as installed. I know they are correctly installed because after going into "[asterisk-source-directory]/channels/h323" and issuing a 'make opt', it correctly builds

Hi, We have been running asterisk in our telco interconnect box with ss7 and H323 configured. Everything ran find till now, however, today, it started crashing with the following messages: [Jul 28 14:56:55] WARNING[2968]: acl.c:541 ast_ouraddrfor: Cannot create socket [Jul 28 14:56:55] ERROR[2968]: chan_h323.c:962 __oh323_rtp_create: Unable to locate local IP address for RTP stream [Jul 28

What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp, ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was not created so no h323 support in asterisk. Just wondering when to expect it again because I was stupid and didn't make a backup of the asterisk code before wiping the directory for a rebuild. -------------- next part -------------- An HTML attachment

When the support for h323plus was announced for Asterisk 1.4.25, I tried to build this support in Asterisk. For this, I checked out the h323plus CVS from SourceForge, which reported version 1.20.beta5, and also the ptlib-2.4.2 source RPM from Fedora 10. I finally managed to build a chan_h323 for Asterisk 1.4.25, which apparently loads correctly, but now I see that I get a segfault whenever I

Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make

I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [asterisk@jonux h323]# make clean rm -f *.o *.so core.* [root@jonux h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN

---------- Forwarded message ---------- Date: Wed, 11 Jun 2003 01:10:16 +0200 (CEST) From: Siggi Langauf <langausd@fachschaft.informatik.uni-stuttgart.de> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go? On Tue, 10 Jun 2003, Jeremy McNamara wrote: > If you would have followed the build instructions laid out by the Open > H.323 folks

I have one and only one reason to use OH323, and that is it works with Cisco's CallManager. The standard H323 channel has one-way audio issues when connected to CCM and fixing them has been identified as a low priority/never going to happen task. Dan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

We have an odd problem, where inbound H323 (chan_h323) calls will sometimes cause a Ringing Congestion that appears to keep the channels open and never release it until we kill and restart asterisk. These "Ringing Congestions" start to pile up, which eventually crashes Asterisk. H323 Gateway -> Asterisk (chan_h323) -> Tor2/PRI -> PSTN Has anyone ran into this problem or

Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h323.conf like: [office] type=h323 prefix=9 context=outbound I get a message saying:

Dear ALL: I find that everybody discuss with asterisk-oh323 instead of chan_h323 of asterisk channels. Why do they choose asterisk-oh323 and not use the build-in module (chan_h323)? What different between these two componments? In my guess, the build-in module should be easy to implement rather than setup another application(asterisk-oh323) expect some bugs or failed functions on chan_h323.

If I let modules.conf autoload chan_h323.so then when I try to stop asterisk, it *does* stop (files in /var/run/asterisk/ are removed and connection via -vr from another console is not possible) but the "asterisk process" stays alive and stalled. In other words, a 'ps -ae | grep asterisk' show that the process is there after running 'stop now'. I either need to press