similar to: chan_oh323.c Segmentation fault during Openphone/Gnomemeeting connect during module loading...

a) jerjers been doing a lot commendable work for * b) support is not mandatory, and i agree with royk it should not be withheld based on political viewpoints, that's pointlessly draconian c) choice is always good, so people should have the option of oh323 or h323, let them decide, and not limit them, unless astmaster chooses to limit them, and that too based on valid points d) jerjer gave a

Hi, I'm trying to get H323 support using asterisk 0.4.0 Unfortunately the pwlib and openh323 versions mentioned in the asterisk-oh323 readme file are no more available, and I had to use newer ones. Now I installed all libraries, but got a segemntion fault when starting asterisk while reading the chan_oh323.conf file. When I declare more than 9 gwprefix I get first a error "out of

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

Sheesh. I only joined here a few days ago and already there's a flame war. Look, to remove your name from the list is easy. It tells you where to go to manage your subscription down there at the bottom. If you want another mailing list, why not go to yahoo!! or topica and set one up, or set one up yourself. It ain't rocket science with mailman. Even an idiot like me has managed it.

faxguy, maybe you can tell me why As I've noted in previous posts I'm evaluating HylaFax with IAXModem. When I run iaxmodem and faxgetty through a console the modem works 100% I have yet to find a fax that it won't tie up with. When I run IAXmodem and faxgetty in initttab, the modem is extremely slow to respond and only actually does anything about half the time, the rest of the time

Hi all! I'm newie to asterisk and I've been trying to make it work in order to use it with Linux softphones (H.323, SIP or IAX, I don't mind) and none hardware phone. I'm using asterisk packages from Debian SID (my distribution), asterisk, asterisk-config, asterisk-sounds, asterisk-h323. I've still not tried with any IAX softphone (gnophone?) but with linphone (SIP) I've

Hi I presently have a line in /etc/inittab (for 5.X based systems) like: v1:2345:respawn:/home/silentm/bin/myfile which runs for any runlevel in 2345. I am looking at migrating that to the new upstart stuff. I can create a file in in /etc/init called myfile.conf and in there have: # /etc/init/myfile.conf start on startup This seems ok... However - my question is how do I stop they myfile

Hello all, I just installed Asterisk with H323 support (chan_h323 from Jeremy McNamara). But experience problem while connecting OpenPhone to Asterisk Here is h.323 trace: 5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720, handle=27 5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225

What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?

I figure that I've successfully loaded and compiled the h323 module into asterisk I can successfully place a call from openphone to a sip phone (snom200) So I figure that the h323 module is working. The question I have is how do I return a call from the sip phone to openphone? I get an error message Sep 7 17:09:49 NOTICE[110992304]: chan_h323.c:861 oh323_request: Asked to get a

Hi, all I am trying to write a function pass to replace a old function to a new function like: int haha(int a) { } ==> int haha(int a, char *IO) { } The below is a part of my code and generate segmentation fault. Can you help me to fix it? Thanks, Shawn // Since we have now created the new function, splice the body of the old // function right into the new function, leaving the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, i tried to configure Gnomemeeting for Asterisk, because its, how it looks, the only tool which gifes me all i want for the use in linux... I have allready installed and running h323 support in asterisk and edited the h323.conf. But i have no chance to configure Gnomemeeting that it connects with Asterisk! I found also nothing useful in the

hello i was searching for solution to problem (sip->h.323). any one from this list asterisk mailing have any idea how to fix it. i am getting error when i try to call from sip to h.323 user i am successfully registering my asterisk box with gnugk. but when i try to call to h.323 openphone on working on GnuGatekeeper, asterisk is not routing it to GnuGk. i am getting the following error. do

Hi all, I have seen many posts on the Shorewalllists dealing with H323. Although lots of them indicated that this is difficult process with kernelrecompilation etc. I just tried what seemed to be logical for me. Surprisingly it worked. Configuration: WS1 ----- FW ------ Internet ------- WS2/Shorewall WS1, FW and WS2 run Redhat9 with its standardkernel 2.4.20. FW and WS2 run Shorewall

Hi All I am having this problem when setting up a H323 call. Can anybody tell me what is going on? Thanks Serge ------------------ NOTICE[245776]: File chan_oh323.c, Line 1293 (oh323_write): H323:1637: Format changed from 4 to 8. WARNING[245776]: File frame.c, Line 76 (ast_smoother_feed): Smoother was working on 4 format frames, now trying to feed 8? ERROR[245776]: File chan_oh323.c, Line 1380

Hi all, if i make a call between one SIP soft-phone to an other soft phone over asterisk, i get a Segmentation fault after take up. The extension is : exten => _00.,1,Dial,OH323/${EXTEN}@<myip>|60|r This means, if a SIP client comes with 00* then dial to <myip> over H323. If the H323 client takes up, a Segmentation fault occures. But, if the extension is exten =>

Hi im running several gentoo servers with Asterisk, only using IAX2 and SIP. Recently we decided to implement h323. All the necessary dependences for oh323-0.7.3 were installed by portage (package manager of Gentoo distro), including openh323, pwlib etc. The module is successfully loaded (load chan_oh323.so) but when asterisk is stopped (stop now) or the oh323 module is unloaded (unload

Hi, I have 2 h323-gateway (clarent-h323 & cisco AS5x00-h323) with different fastStart/h245Tunneling mode configuration. I'm using fedora core 1 with - asterisk-1.0.9 - asterisk-oh323-0.6.6 Anyone know how to configure oh323.conf with multiple h323-gateway ? regards, __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten => 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work).... Thanx in advance, Victor...