similar to: Difference between native and 3rd party h323 channel driver ?

Hi, I have downloaded files and also local versions of pwlib oh323 (both Janus patched). Both libraries compile fine, but I get following errors on asterisk-oh323-0.6.5. Readme is a bit confusing since it doesn't mention which local libraries should be downloaded from inaccess to get everything working OK. I've also tried with/without patching oh323 with supplied patch. Any hint, advice

Hi, anybody knows what's missing or problem why i cant compile asterisk-oh323 in my machine? i got this compiled successfully ...Openh323 - v1.12.2 ...pwlib - v1.5.2 except ...asterisk-oh323 - v0.6.5 here's the output as i run make... mkoy@sambag:~/voip/asterisk-oh323-0.6.5$ make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory

Hi there ! I searched the whole web to find some helping information about H323 Control Protocol, but there is no way to find that information. We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2 + 'asterisk-oh323_1.5 channel driver + wrapper' and configured the dialplan for using our H323 Endpoints which are ip200 Innovaphones. Besides, we also use Gnomemeeting but

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

I m new to asterisk n i've got an IP phone that supports h323 protocol.... but i dont know how to configure asterisk to use it... i m comfortable in using sip & iax softphones.... but there is no h323.conf in /etc/asterisk/ .... i read that i've to compile some files but i m confused regarding h323 & oh323 ...... which one should i use.. plz tell me or atleast give some helpful

i read that asterisk supports iax,sip and h323 protocols.... i've used sip & iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 ..... i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable to place calls to/by my phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf

Hi, I'm trying to install asterisk with h323 support on rh9 box. I want to find a working combination between asterisk,asterisk-oh323,pwlib and openh323. Thank you! Ginel Tudorache

I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. Firstly, despite the warnings in h323/README, I decided to try using the distro-specific versions of openh323 and pwlib. Of course, the Makefiles in channels and channels/h323 assume that openh323 and pwlib have been specially compiled in $HOME, so I modified the Makefiles to look for headers and libraries in

hi all im trying to compile h323 i have got the pwlib and openh323 working that is simph323 is running properly but when i try to compile h323 in the channels directory it gives me the following error can anybody please help me with [root@test src]# cd /usr/src/asterisk/channels/h323/ [root@test h323]# make opt g++ -DNDEBUG -I../../include -Wmissing-prototypes -fPIC -DP_LINUX=2.6.5-1.358

Rafael J. Risco G.V. wrote: > > On Sat, 11 Dec 2004 16:49:12 +0000, Corvin <corvin.dun@wp.pl> wrote: >> Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa?: >> > Hi. >> > >> > I need to set up H323 on an Asterisk box. I've succesfuly compiled the >> > asterisk oh323 (including of course all the dependencies: PWlib and >> >

i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there I am pretty close on giving up on Asterisk :-/ I am (still) trying to make a call from a H323 phone to an Asterisk provider using AIX. But H323 does not route the number to AIX. All it is transmitting is an "s". *CLI> -- Executing Dial("OH323/R27865", "IAX2/demo:demo@gw1.musimi.dk/s") in new stack -- Called demo:demo@gw1.musimi.dk/s Jul 2

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

AVE! i am trying to register h323 asterisk to the gatekeeper as i installed asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323 on fedora core3 on a cisco mcs 7800 server problem is i want the asterisk to register with gatekeeper endpoint with specific zone name and type... i searched the web, mail list but there weren't any helpful ones could anyone plz tell me how to

Hi, I start to be a little confused so I am asking to the list. I want to make with * a gateway from PSTN to H323, and to send all incomings call to a predefined IP, which will treat the h323 calls. let's assume that all my incoming numbers starts with 00 here is my extensions [incoming] exten => s,1,Answer exten => _00.,1,Answer exten =>

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No

Hi Mark Funny you should ask this question, I just spent yesterday integrating building asterisk with h323 support to connect to a Cisco call agent.....I cant say if it will work for you but it compiles and loads nicely ! I will be testing this evening.... # cd /root # wget http://www.voxgratia.org/releases/pwlib-Pandora_release-src-tar.gz # wget

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some