similar to: Routing incoming H.323 calls to specific contexts.

hi all I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h software can only be configured to point to a gatekeeper. I know I don't need to do this, but it's a test before I setup my Symbol Netvision phones (I don't have an access point for them now). First, this is what I have, and what

Hi, I've got two H.323 Client connected to Asterisk, when one of them requests boeing connected to the other I use CALL application and both get in touch trhough asterisk, but using Call Asterisk stays on the middle and the sound quality gets poor. Is there any way to 'transfer' the call so Asterisk doesn't stay in the middle ?? I use OpenH323GK (www.gnugk.org

Hi, - I have account on VoIP provider sample.net with auth for example Username: test and Password: test - this provider VoIP use only H.323 and I must use softphone with H.323 support to make phone to PSTN via this provider My idea is configure asterisk to possible make phone from my users SIP softphones to PSTN via this provider sample.net. - I have asterisk with support H.323 ( Ihave use

Hi, we intend to use innovaphone's IP200 full standard compliant H.323 hardware phones (great!) with asterisk. Everything works perfect (calls, signalling, talking) until we want to perform a call transfer (H.450). Upon pressing the R-key the first connection is hung up immediately. As gatekeeper we use a IP400-box OR GnuGK, same result with both... So, who has experience or uses

we are trying to use asterisk for converting SIP to H.323 calls. asterisk (0.9.1) runs on the same linux (Redhat 8) box of our gatekeeper (gnugk version 2.0.8). the calls are going out through a cisco gateway. when I make a call from a SIP phone to a PSTN number reachable through the cisco gateway: asterisk diaplays Aug 5 23:24:26 WARNING[1255648560]: chan_oh323.c:2898

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i have a working Microsoft ISA firewall with buildin H.323 Gatekeeper.... So Far, i got registerd the asterisk on the M$ Gatekeeper... here is the h.323 configuration: ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs dtmfmode=rfc2833 gatekeeper =

Hello everybody, I?ve been pulling my hair for a week now over a problem, and I really don?t know where to look anymore. Here?s my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323 and openH323, and on the same server I also setup GnuGK (10.253.30.1). I

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and

Hi everyone I have successfully compiled and installed OH323 support (finally) into my Asterisk. I want to connect the Asterisk server to our Alcatel OmniPCX Office (OXO) PABX, which has an internal H.323 gateway. I have created the correct dialplans in Asterisk and same in OXO. The OXO only supports G711a G711u G729 and G723.1 codecs. When I call from a SIP phone to OXO using my

Hi, I am working on a H.323 project which involves GnuGk and Asterisk My current goal is to provide IVR functionality for the H.323 users which register through GnuGk(eg. call credit information) I have successfully built a H.323 platform using GnuGk - it uses SQL accounting and authorisation. Now I am trying to integrate it with Asterisk in order to provide IVR functionality as I already

Current setup: Polycom IP3000 <-> gnugk <-> asterisk <-> Cisco 7940 Asterisk and gnugk are on 10.20.98.6 IP3000 is H.323, using G.711 (10.20.98.2) 7940 is SIP, using g711ulaw (10.20.98.3) I've been asterisk for a while now, only using SIP devices. I'm happy with that side of things, but I've not used H.323 before this week, in trying to get the IP3000 to work. *

I have 2 FXO channels from which I want to route incoming calls to different contexts in extensions.conf. I edited the context entries in dahdi-channels.conf and created matching entries in extensions.conf. One channel is routed to the new context as I want, but the other channel is stuck going to the default "from-pstn" context no matter what I do. Can anyone see what I've

Hi, I'm using asterisk with chan_h323 together with gnugk. chan_h323 and gnugk were recently compiled with pwlib-1.5.2 and openh323-1.12.2 as advised. When connecting asterisk directly by ohphone (without gatekeeper), everthing is fine. When using gnugk for usage control in routed mode, I find a funny situation in asterisk's H.323 debug: == New H.323 Connection created. --

With the lack of info on Yoda Communications in Taiwan and their hardware, I thought I'd post my experience. I got my hands on a few H.323 VG-400's and VG-100TA's. http://www.yoda.com.tw/model.php?type=VoIP_Solution&pname=VG400 2 of the VG-400's were 2FXO/2FXS models. A couple were deployed to SE Asia, where we planned to offer our services. Originally, I ran a GnuGK server

Hi! I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323 built with needed PWlib v.1.5.2 and open H.323 v.1.12.2. But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's compiling fails and I get error 1. Do you have any working solutions with asterisk and

Hi, i'm trying to setup Asterisk to work with GnuGK using the Openh323 channel driver. I have a Gatekeeper that gets H.323 calls from a Cisco GW. To this Gatekeeper I've registered some endpoints, Cisco ATA186, Snom 100, etc. Now i want send the numbers 083xxx into Asterisk. Easy, i'll just enter something like this into oh323.conf: gwprefix=083 And all my calls starting with 083

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form

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

I'm trying to get a Snom100 configured with H.323. Right now, the phone is not even connecting to the Asterisk server, so there's obviously a problem with the snom config. Does anybody have a sample working configuration with the snom phone, using H.323? I've checked the archives, but everybody seems to be using SIP with the Snom phone, not H.323. -Tilghman

Stephan, Ok, I'll re-state the problem... I have two devices that I want to talk to each other: 1. an Asterisk PBX 2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk) both devices are effectively "gateways" because they have many subscribers behind them. The Damm Cellular system controller is based on Windows-XP Embedded and its sub-systems used the OpenH323