similar to: Re: chan_oh323.c ast_oh323_new Internal channel initialization failed

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

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

hello i want to rout my calls to h.323. i have registered my asterisk with GnuGatekeeper. but it is not routing my call to h.323 channel. he is saying Internal channel initialization failed. Bad binary? can any one check my settings what is problem here thanks in advance kamran exten=>_321XXXX,1,Dial(OH323/${EXTEN}@192.168.0.153:1719,30,r)

My apologies if this question has been answered previously. However, I found that it was nearly impossible to search and find since anything can cause a segmentation fault. Problem. When Asterisk is booting up the h323 modules and a client tries to connect before Asterisk/h323 is finished booting, the program seg faults out and doesn't load. I thought about putting this into the inittab,

Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. " WARNING[8446]: chan_oh323.c:3555

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; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

Hi guys, I have Rails 3 running on Windows 7 with Ruby 1.9.2 Every now and again, my server crashes, and I see the following information in the windows event log: Faulting application name: ruby.exe, version: 1.9.2.180, time stamp: 0x4d5ee5ed Faulting module name: msvcrt-ruby191.dll, version: 1.9.2.180, time stamp: 0x4d5ee5ec Exception code: 0xc0000005 Fault offset: 0x0011a00e Faulting process

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 any one tell me how to make a dialplan my extensions.conf exten => _40XXXXXXXXXXXX,1,Dial(OH323/${EXTEN}) i want to dial to 40XXXXXXXXXXXX number. XXXXXXXXXXXX could be any number like 923335224005 or 92512213248 at the moment when i am trying to dial 40923335224005 asterisk is dialing Executing Dial("OH323/R11429", "OH323/40923335224005") but i want him to dial

Hi Everyone, New to the list. Thanks in advance for any assistance you are able to provide. I have a TrippLite SMART2200RM2UN UPS. I have installed and configured NUT as instructed on the website, and am able to monitor the status of the UPS without much problem. The only problem I am seeing is that I cannot get the machine to actually send a Low Battery ( LB ) signal. When I run

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

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 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

hello like if 6000 is the main exchange number. any one dial to 6000 will be asked for pressing his desired extension then he can press his desired extension then his number is diled exten=>6000,1,Background(enterdesiredexten) exten=>6000,2,Wait(2) exten=>2000,1,Dial(SIP/${EXTEN})

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

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

> > battery.charge: 3 > battery.charge.low: 10 > battery.charge.warning: 30 > battery.runtime: 93 > battery.temperature: 32.9 > battery.type: PbAC > battery.voltage: 46.4 > battery.voltage.nominal: 48.0 Is it possible that battery.charge is really 30% rather than 3%? The 3016 protocol models have issues with scaling on some of the voltages and frequencies. You can see

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 all, this is my actuel setup [SIP 2005]--[asterisk]--H.323 Trunk--[PBX]--[ext. 8900] Linux CentOS 3.3 (2.4.21-20.EL.c0) asterisk-1.0.1 asterisk-oh323-0.6.3b openh323_1.12.2 pwlib_1.5.2 Calling from SIPphone to the extension 8900 works always. Calling from 8900 to SIPphone works only sporadicly without any recognizeable pattern. Find below the output of the debug command: asterisk