similar to: H.323 security flaws article

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

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

Hello, Good news on the Polycom front for those that are interested. It looks like we may get a dedicated Engineer for Polycom/Asterisk!!! Happy Day! Here's the message I got tonight: Matt: I heard back from our VP of Engineering- she is prepared to have an individual dedicated to working on the Digium- Asterisk project. Can we discuss again Friday or mid next week? Scott Willard

Hi all, I'm a student (my thesis work consist in testing interopearbility SIP/H.323) and I begin to work with asterisk in this days. I have to testing to SIP/H.323, since today I have used Vocal system, but there are some problem for this features. In the asterisk mailing list, in the next message I've seen an e-mail """ [Asterisk-Users] Cisco

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

Hello, I'm running Asterisk 1.0.1 (the same was with 0.9, 1.0). When it receives inbound H.323 call it makes connection and uses local 127.0.0.1 address to send audio stream: remoteIpAddress: 127.0.0.1 When making outbound calls from Asterisk it makes correct connection to send audio stream. Is it a bug in h.323? Is there some more settings to make in .conf files? See detailed debug below:

I'm new to H.323 and I have noticed that there are two separate channel drivers for * available - the inbuilt one, and oh-323. I had trouble compiling oh-323 with the current cvs stable, so I tried the inbiult one (with specifiec recommended versions of openh323 and pwlib). It compiled cleanly but I am told that it is not recommended (unstable?). Can someone with first-hand * H.323

Hello, I'm trying to connect Cisco Call Manager 3.3 with Asterisk using H.323 Gateway. When I place call from a SIP phone registered at Asterisk to SCCP phone at CCM I can hear the voice in both directions. But when I call from SCCP phone at CCM to SIP phone at Asterisk the voice goes from CCM to Asterisk only. All devices have real IP-addresses - no NAT is used. Asterisk console does not

I'm having trouble registering my asterisk to a cisco gatekeeper. I do not have control over the gatekeeper, and I know that it has user info defined in an LDAP. I have a user name and a password that I can use, but I can't seem to get Asterisk to register on the gatekeeper. I can't find exactly how I'm supposed to define the gatekeeper in the h323.conf file. This is the response

I''ve got a brand Polycom Viewstation FX video teleconferencing unit. I''ve got a Shorewall firewall 1.4.9 instance running on box with a 2.4.20 kernel. I can not receive H.323 video. From the FAQ I''ve read that there is an unsupported H.323 connection tracking module that is no longer maintained. What options are there to make H.323 work with a IPTables based

Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs

all, How amenable is Asterisk to a setup that looks something like this? { SIP-only VoIP hardphones } <===> { Asterisk } <===> { Cisco H.323 gateway } <===> { trunks to PSTN } I've heard Asterisk didn't play too well with H.323, but I wanted to get some more details on that. I only recently completed my first Asterisk testbed, using four softphones and an Asterisk

I've got a problem connecting to Cisco call manager. I dial a numder and hear only ringing h.323 debug shows this Allowed Codecs: Table: G.711-ALaw-64k{sw} <1> Set: 0: 0: G.711-ALaw-64k{sw} <1> -- Making call to 3200@10.1.105.3. == New H.323 Connection created. -- 6129 is calling host 3200@10.1.105.3 -- Call token is

Hi, I am trying to make a call from SIP to H.323 using chan_h323. Asterisk CVS-HEAD-03/10/05-10:08:22. As given in chan_h323 readme I compiled pwlib and open h323 versions 1.8.1 and 1.15.1. Call seems to be get connected but no audio path. I see following; -- AGI Script Executing Application: (DIAL) Options: (H323/YYYY#XX112422428@XX.103.19.91/XX112422428|60|HS(63840)) -- Setting call

Sheesh. I only joined here a few days ago and already there's a flame war. Look, to remove your name from the list is easy. It tells you where to go to manage your subscription down there at the bottom. If you want another mailing list, why not go to yahoo!! or topica and set one up, or set one up yourself. It ain't rocket science with mailman. Even an idiot like me has managed it.

I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323. Trying to find a way I could use my laptop to send a fax over H323 to the BrookTrout card for testing. Any thoughts? Normally I'd setup a FXS interface on a Cisco router and setup a h323 dial peer to the BrookTrout, but I didn't the router with me! ----------------------------------------- Disclaimer: This

We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... " I am sorry that's not a valid extension" before i get a chance to enter anything. Any

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 ______________________________________________________________________________ Caldera Systems, Inc. Security Advisory Subject: flaws in the SSL transaction handling of Netscape Advisory number: CSSA-2000-017.0 Issue date: 2000 June, 09 Cross reference:

Hello, Just confirmed this on my end, because of the massive changes that have been made to callerID handling in asterisk 1.0.5 many of the features of the astGUIclient suite will not work on this new version. The latest stable version recommended is Asterisk 1.0.3. We will work on trying to find ways around the new callerID rules that the asterisk developers have put in place and hope to have

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call