similar to: H323-to-SIP proxy

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.

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

Hello everybody, First time installer and I need the lists advice. My plan is to use asterisk PBX with some hardware to terminate my calls coming from several operational gnugk gatekeepers. Do have RH9 and downloaded the latest asterisk from CVS. Compiled according instructions and is running fine. Could hardly find any info on h323 implementation untill the REAME in the channels directory.

I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4) The pwlib + opal packages don't satisfy Asterisk's configure script (to let H323 compile), so I removed those and added the latest ptlib + h323plus (from h323plus.org) I can compile ptlib and h323, but when I load chan_h323 in asterisk I get a segfault. I had to point LD_LIBRARY_PATH to /usr/local/lib with the

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

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 >> >

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

Hi all Asterisk Gurus. I am really badly in need of help. Asterisk is very lovely software, but has one big disadvantage.. lack of documentation.But let's get to the point. 1. Is it normal that I get such a crappy quality with iax, some drops and clicks? Could anyone with some similar setup check my quality and say if this is what the people are so excited about? ( I used to work as a speech

The following occurs with code from yesterday's cvs (asterisk) and current OpenH323 code: [root@raid-2 h323]# make clean install rm -f *.o *.so core.* cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations - DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIA N -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING

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, what's the difference between h323 and oh323 modules? which one should I use? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030722/3d3edb73/attachment.htm

Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Aug 10 09:09:18 WARNING[7824]:

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

Hello, I am moving topic from asterisk-dev list to asterisk-users list. Did anyone succeed receive incoming calls in h323 and orient them to right context based on "host" identification? To summarise, I have quintum Gateway sending call to Asterisk box, and I would like to use asterisk as a protocol converter h323 --> sip. in h323.conf, I have [quintum_gw1] type=user

Hello all, I want to use asterisk as protocol converter, H323 to SIP. I am using Asterisk 1.2.14 with chan_h323 and the free version of g729. When calling from SIP to H323 everything is fine. But when calling from H323 to SIP, the phone using SIP doesn't hear the other party. The phones and Asterisk are in the same subnet and the firewall of the Asterisk box is off. Please advice. Thank you,

Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more,

Some of you may remember seeing my issue using SIP for incoming calls from the PSTN: http://voip-info.org/wiki-Asterisk+cisco+FXO i.e. all incoming calls arrive in the default 'bogon-calls' context. Well, I tried again using H.323 & get exactly the same result (both for chan_h323 & chan_oh323) i.e. all attempts to put a type=peer in sip.conf or a type=user in h323.conf for

Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

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