similar to: h323 with innovaphone ip 400 gatekeeper/innovaphone Ip200 phones

Hello, I think I have the same problem as Martin Bene mentioned in http://lists.digium.com/pipermail/asterisk-users/2004-January/034521.html Since I found no further information about this I'd like to ask wether you know what the reason for this problem is and how one can get around this. * is registered to the innovaphone gatekeeper. Trunk connection is done with an AVM-B1 and chan_capi.

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

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

Hi all I am trying to compile Asterisk on RH9 I am have installed pwlib_1.5.2 readline-4.3-5 readline-devel-4.3-5 openssl-devel-0.9.7a-2 openssl-0.9.7a-2 openh323_1.12.2 I read that it has to do with pwlib not being installed correctly so i made sure it was in /usr/local correctly When i try to compile i get the follow errors.... I was wondering can anyone shed in light on this issue Thanks

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

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 all I am trying to compile Asterisk on RH9 I am have installed pwlib_1.5.2 readline-4.3-5 readline-devel-4.3-5 openssl-devel-0.9.7a-2 openssl-0.9.7a-2 openh323_1.12.2 I read that it has to do with pwlib not being installed correctly so i made sure it was in /usr/local correctly When i try to compile i get the follow errors.... I was wondering can anyone shed in light on this issue Thanks

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

Hello, when using BLF with Asterisk 1.6, I notice that the Caller-ID information is not displayed on the monitoring key of my Innovaphone IP200A. If the IP-phone of my colleague rings, I should see on my partner key the number of the caller. This is information that is being send in the xml-body of the NOTIFY-message. I do not see this information in the xml-body of a NOTICE-message from

Hi all, this is my actuel setup [SIP 2005]--[asterisk]--H.323 Trunk--[PBX]--[ext. 8900] Linux CentOS 3.3 (2.4.21-20.EL.c0) asterisk-1.0.1 asterisk-oh323-0.6.3b openh323_1.12.2 pwlib_1.5.2 Calling from SIPphone to the extension 8900 works always. Calling from 8900 to SIPphone works only sporadicly without any recognizeable pattern. Find below the output of the debug command: asterisk

We're using a couple of h323 IP Phones (innovaphone ip200) w/ asterisk. Basic call setup works, but we can't get call transfers to work: On pressing the "transfer" button on the phone (getting a new dialtone) the 2nd endpoint is disconnected. Any idea if we can get this to work? Same reaction using the innovaphone ip400 gatekeeper and using gnugk. Asterisk version is 0.7.2

Hello.... Asterisk is compiled and running perfectly... But when i try to compile Channel_h323... that's another story :-|. I'm compiling using RH9, OpenH323_1.12.2, pwlib_1.5.2, i compiled and installed them myself... GnuGk is running smoothly using them. Also my intention is to terminate calls from a SIP client through Asterisk and into my GnuGK... has anyone accomplished this ?

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

I have asterisk connected to E1 interface with Digium TE110P. I have Cisco ATA 186 and fax pass thru well. I have tried to establish modem connection (from computer connected to ATA => SIP => * => E1 => Telco => pstn => another modem) and I do connect (at 14,400) but connection end after a minute. How to establish successfully modem connection? Has anybody tried innovaphone IP21

> > Does anyone know of Phone that supports G.723 on H.323. > Innovaphone tiptel 200 for example. http://www.innovaphone.com/webneu2/products/en_IP200.asp One of the nicest phones I've seen so far, h.323 only though. Bye, Martin

Hi all, I have one asterisk server with one ISDN BRI connection to PSTN, with h.323 support (oh323) I buy some voip phones, and I connect them to the same switch as asterisk server is; all is at the same TCP network. I need to route some extensions from my DDI (DID) line at asterisk to some voip phones, and also to do: outgoing calls from any voip phone to PSTN via asterisk, and intermediate

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

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

Hi, I just got some interesting information from Kirk Telecom (www.kirktelecom.com). This company has been in the business of providing DECT solutions (IP gateway, base stations, repeaters and handsets) either to be used with Cisco CallManager (SCCP protocol) or with the Innovaphone IP PBX system (H.323). Two important elements: 1. It seems they foresee a SIP version of their product in Q1

hi, i had asterisk and gnugk running on fedora core 2. it worked quite well. then, i needed to change to red hat 9, and i'm experiencing troubles with h.323 :-( making a call from a h.323 phone (innovaphone) does not work, and dial-in also doesn't. below is an excerpt of what happens, when i try to dial-in my extension (126). it takes about 10(!) seconds, until the 'Called 126'