similar to: chan_oh323 delay

Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi; Where I can find Caller IP adress or caller Login in chan_oh323.c (asterisk-oh323-0.5.5) ? I try "cd.call_source_alias" but I don't have this. Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030908/bbf65ae9/attachment.htm

hi everybody, Is there SIP client which work with Asterisk and can be embedded in a HTML page ? Thanks Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/226a8b1b/attachment.htm

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

we got this installation : WinSip(demo version) -> ser(radius accounting) -> asterisk(from sip to h323 channel) -> gsm gateway(with 32 sims in it) we configured winsip to make 28 calls like from 28 different sip accounts, to 28 different cellular phones numbers after the first ten : -- Executing Dial("SIP/5060-081925b0", "OH323/33xxxxxx@gsm.gateway.ip") in

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 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 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 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, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030804/af4203ca/attachment.htm

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, 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...

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'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

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):

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

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

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

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

hey people, i've installed an asterisk version on my Linux Distribution. Then i wanted to compile the packages for H323 support. The openh323 and pwlib libraries compiled without any error. But at least i got one in /~/asterisk-oh323-0.5-10/asterisk-driver/chan_oh323.c in Line 1128. Too many aruments caused this error. Is anybody able to say why and how can i solve thsi problem? Kindly Regards