similar to: compile error in chan_oh323.c

Hello, i have some problems with compiling oh_323 (0.6.1) and asterisk (0.9). I successfully have compiled the necessary libs pwlib and openh323. I have set all path-variables in the oh_323 makefile. When i compile the oh_323 channel driver i'v got some errors. in function oh323_call chan_oh323.c: 1135 ... too few arguments to function 'ast_queue_hangup' So what could be the

Hello, i want to use asterisk as a gateway for H323-Phones. But i cant get it work. I'm using a gatekeeper on another computer. My IP-phone is registered there. Does anybody can sent me an oh323.conf and extension.conf as examples? Thanks in advance Erik Bastian -- NEU : GMX Internet.FreeDSL Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl

Thanks, but I am still having the same problem -- I made sure to reference the correct passwd file, but WinXP is still forcing me to logon as AGAMEMNON\Guest. Tim -----Original Message----- From: miles123@gmx.de [mailto:miles123@gmx.de] Sent: Monday, May 03, 2004 6:11 AM To: tbbooher@cox.net Subject: SOME Suggestions on [Samba] Can't the very basic stuff to work Hello Tim, I have read

Hello! I need to test the adjustment of a (Negative) Exponential Distribution to a dataset. The parameter of the distribution is unknown. What is the appropriate test to do? I've tried the ks.test, although I think this isn't the appropriate one, as I don't know the parameter. Can anybody help me? Thanks in advance, Janete -- Janete da Silva Borges janeteborges at gmx.net Ab

I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what

Hi, > chan_oh323.so: undefined > symbol: __use_ast_pthread_create_instead__ is not a bug, it's a hint: use "ast_pthread_create" instead [what your were using] and means: replace in asterisk-oh/asterisk-driver/chan_oh323.c at line 3764 "pthread_create" by "ast_pthread_create" Roger.

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 all, I want to install h.323 support for *, but when I launch * from shell command asterisk -vvvc I have the next error screen: [chan_oh323.so]WARNING[1024]: File loader.c, Line 226 (ast_load_resource): liboh323wrap.so: cannot open shared object file: No such file or directory WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so

If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I have asterisk version 1.0.0,have comipled pwlib version 1.5.2 ;and openh323 version 1.12.2.When I try to build asterisk-oh323 version 0.5.9 or 0.5.10 ,I get the following error : make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 I thought that there might be some linking problem,so I searched

Hi, I have asterisk connected to PSTN via H.323 gateway via chan_oh323. Incoming calls to SIP extensions work, but SIP message "486 busy here" from a busy extension isn't correctly forwarded to H.323. As a result, a caller from the H.323 side calling a busy SIP extension gets some rings and then an irritating timeout with H.323 message 'no user responding' instead of

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

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

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 have a communication partner who needs G.729. When I disable in oh323.conf all codecs except G.729A, everthing is fine, except that I can't receive calls with other codecs from other partners via H.323. That's why I enabled also GSM and G.711 in oh323.conf and put a SetVar(OH323_OUTCODEC=g729) in my extensions.conf for that partner. Now I see in the console log: -- Executing

I am trying to figure out the correct way to send my E.164 ID with chan_h323 and or chan_oh323 as my H323 provider requires this in the format of 'account-pin'. With chan_oh323 I have been able to register with the gatekeeper and can recieve incomming calls, but outgoing calls do not work. With chan_h323, I can call H323 clients (netmeeting, ATAs etc) but cannot place a call through my

---------- Forwarded message ---------- From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> Date: Oct 4, 2007 12:56 PM Subject: Re: [asterisk-dev] chan_h323 and chan_oh323 compatibilities To: asterisk-dev at lists.digium.com Hi On Thu, Oct 04, 2007 at 11:46:30AM -0300, Caciano Machado wrote: > I'm receiving a lot of warning messages from my Asterisk > 1.2.5/chan_oh323 every time

Hello I have the following problem using chan_oh323 I have DialGate 2160 for SystemBas (www.sysbas.com) connected to PSTN (H.323 to FXO/FXS gateway) when i try to make call form one pstn phone to other trough asterisk or when i make call from software h.323 client trough asterisk and this gateway to pstn i have the problem with voice quality. The side that initiated call can be heared clearly,

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

hello i try to call from sip phone on asteris to open phone on GnuGK. can any one tell me why it is saying chan_oh323.c:2501 ast_oh323_new: Internal channel initialization failed. Bad binary? Mar 16 13:28:46 WARNING[5963]: chan_oh323.c:2727 oh323_request: Failed to create new H.323 private structure 4. Mar 16 13:28:46 NOTICE[5963]: app_dial.c:749 dial_exec: Unable to create channel of type