similar to: h323 to sip ringing indication

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,

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

Hello all, I am trying to use native h323 built from asterisk 1.2.7. I configured the h323 to receive incoming calls...the problem is i can receive the call to my asterisk and it rings another extension but no audio. I don't see any good documentation about gatekeepers, fast start, etc with h323. I would like to get some help from you guys to fix this issue. If any of you have configured

Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make

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

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

I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I

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,

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

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

Greetings to All, I hope someone has already gotten this working. I spent all day today trying to get ooh323 and gnugk to run on the same box. After a lot of tweaking to get everything compiled, I got both up and running. I can make calls IAX to H323, but cannot make calls in the reverse direction. I have tried many different configs on the GK, but always come up with the same error. It appears

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through

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

Our VoIP provider wants to use a restricted set of ports for H323. All their current users have hardware at the customer premises which does this successfully for them. We are trying to set up Asterisk to do the same (with their co-operation). I am not familiar with the H323 terms for these connections, so bear with me... The (Connect/Listen) Port is 1720 (set in (o)h323.conf) Tcp Range

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 =>

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

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 >> >

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

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

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1