similar to: chan_oh323 loading error

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Colleagues- I am working on the design of a fairly large samba deployment, and I am looking for feedback on some of my design ideas. I have 10 buildings spread out in and around a city, all interconnected via 1.5Mb leased lines. There are samba servers in each building. I have some users that move from building to building. We are using primarily windows 98 desktops, with a few 2K and XPP

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 All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian

Hi all I have asked this question before but have not got any helping input. I'm really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to

Hi, I having a problem compiling the wrapper for oh323. I am running Debian, kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the openh323 version I have is 1.13.5. I execute the following commands first before attempting to compile the wrapper: pwlib_1.6.6: make both openh323 1.13.5 ./configure make opt asterisk-oh323 0.6.2 make

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'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

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

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

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, 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...

After some more analysis of my "dropped fragment" problem, things look like this: Cisco 7940 phone -- RTP --> chan_oh323 --> Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one

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

I'm trying to install asterisk PBX with openH323 support. I installed all the packages ( Pwlib, openH323 and openH323 gatekeeper) from source successfully. i also installed the wrapper ( http://www.inaccessnetworks.com/projects/asterisk-oh323 ). However when i try to start asterisk i get the following errors... ARNING[1024]: File loader.c, Line 212 (ast_load_resource):

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,

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.

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

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

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference