similar to: chan_h323 doesn't pass audio before call is answered

I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration

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:

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

Hi, Has anybody managed to get callerid properly set on a call from local to asterisk SIP endpoint through h323-pstn gateway to a regular phone. I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it. When I place a call to pstn I'm not receiving 12125551234 as the clid, but a number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other

hi IIRC, Jeremy once said that chan_h323 could be used as a gatekeeper but perhaps lacking a few features as compared to gnugk. Is this possible? I have some dlink DPH-100H phoes here for testing, but they require a gatekeeper, and if I can do it, I'd love to keep gnugk out of this. thanks roy

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

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.

Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic

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

Hi, I'm using asterisk with chan_h323 together with gnugk. chan_h323 and gnugk were recently compiled with pwlib-1.5.2 and openh323-1.12.2 as advised. When connecting asterisk directly by ohphone (without gatekeeper), everthing is fine. When using gnugk for usage control in routed mode, I find a funny situation in asterisk's H.323 debug: == New H.323 Connection created. --

hi all I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h software can only be configured to point to a gatekeeper. I know I don't need to do this, but it's a test before I setup my Symbol Netvision phones (I don't have an access point for them now). First, this is what I have, and what

Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -vvvvcd). In debug log I can see the

hi, i had asterisk and gnugk running on fedora core 2. it worked quite well. then, i needed to change to red hat 9, and i'm experiencing troubles with h.323 :-( making a call from a h.323 phone (innovaphone) does not work, and dial-in also doesn't. below is an excerpt of what happens, when i try to dial-in my extension (126). it takes about 10(!) seconds, until the 'Called 126'

Current setup: Polycom IP3000 <-> gnugk <-> asterisk <-> Cisco 7940 Asterisk and gnugk are on 10.20.98.6 IP3000 is H.323, using G.711 (10.20.98.2) 7940 is SIP, using g711ulaw (10.20.98.3) I've been asterisk for a while now, only using SIP devices. I'm happy with that side of things, but I've not used H.323 before this week, in trying to get the IP3000 to work. *

Hi! I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323 built with needed PWlib v.1.5.2 and open H.323 v.1.12.2. But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's compiling fails and I get error 1. Do you have any working solutions with asterisk and

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,

Hi, I also had some problems using chan_oh323 together with gnugk. * <-> gnugk <-> h323-phone When I called the phone and hang up, befor the phone was picked up, the h323-phone continued ringing. The same, when the h323- and some sip-phones were called, and the sip-phone picked up the call first. (It is annoying, when you are talking to someone at the phone and the phone on the

Hi all, I am using chan_h323 from Asterisk CVS to interconnect with GNUGK v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on Asterisk. However, I only heard ringing when the call was answered on SIP side. Below is the debug from chan_h323. Any help is welcome. Thanks. *CLI> == New H.323 Connection created. -- Setting up Call -- Call token:

Jeremy, In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution. Could you post those docs in your download directory? I'm trying to understand the nuances of your driver, gnugk, and extensions. Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content