similar to: is there a variable for the calling IP ?

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,

Dears; I am having same problem, that when I place a call from the H323 end point (even if it is not added in the ooh323.conf), then asterisk handle the call and play the wave file in the default context. Also I added endpoint to the ooh323.conf and same thing, it keep goes for default context whatever the context placed. My Asterisk vesion is 1.4.25 My Asterisk add-on version is: 1.4.8 What I

I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From

I use asterisk 1.2.5 and h323 that comes with addons 1.2.1. Incoming call comes on h323 trunk. Person A (local SIP phone) receives call and tries to make attendant transfer to person B (local SIP phone). They speak. Then A hangs up. Call form h323 trunk doesn't get to person B. This is what I get on CLI. -- Attempting native bridge of OOH323/xxx.xxx.xxx.xxx-5381 and SIP/307-5663 --

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

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

hello all i'm using ooh323.so module for my h323 connections and it works fine. i just have problem with loading and unloading module. you know, ooh323 module doesn't support reload command. it means, if ooh323 module is loaded and i reconfigure my h323 channels (add another channel), i should unload and load module again. it causes to disconnect all h323 connections which are connected

All, I'm running (attempting to) ooh323 with Asterisk and a Cisco 2621XM router operating as a H.323 GateKeeper, however when I bring the Asterisk box up it registers successfully with the GateKeeper (exchanges GRQ/GCF, then RRQ/RCF) it notes the GateKeeper supports keepalive at 300 seconds, when it gets to time to re-register its sends an RRQ again and gets rejected with RRJ (unspecified

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

All, I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working. I have the following: [Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323] 192.168.1.5 192.168.1.6 192.168.1.7 the Asterisk box has numbers

Hello all, I have seen some people asking how to configure asterisk to work with h323 but i did not manage to do fix it yet (i am not an asterisk expert). Can someone help me configuring asterisk? It is already compiled asterisk 1.4.5 with H323 support. Everything looks fine. Then i understand i need to configure several files: -sip.conf -ooh323.conf -extensions.conf do i also need to configure

Dears; Now using Asterisk H323 (which coming with Asterisk, I just compiled PWLIB and OPENH323), now I am placing a call from the IP Phone, the call comes to Asterisk, and it goes to the default context, but did not hear any voice of the played wave file. 1) Why Asterisk accepted the call without authentication? At least, it should be added to the h323.conf. 2) In case we found the method to

hello Dmitry thank you for your reply. Ok, you are right. i want to configure trunk h323 between asterisk 11.13.1 and 2800 cisco router. this is my scenario: PBX(100)--->cisco--->asterisk11.13.1---->PBX(200) when i call from 100 to 200, everything is ok but when i call from 200 to 100, phone rings but after i answer it, i have no voice and call terminates after 5 seconds. this is

This is third time today that my Asterisk hangs up. It seams that I have problems with h323. I'm using ooh323 from Asterisk add-ons. I have the following configuration Asterisk 1.2.1 Asterisk-addons 1.2.1 Fedora Core 4 I'm using SIP phones and h323 trunk to my VoIP provider Like I said this is third time today that he hang's up. First time, I came at work and Asterisk was down.

Can someone tell me how I can configure ooh323.conf to accept call from h323 gateway (only the authorized h323 gateway) to my asterisk. I will be glad to know how this can be done. I tried the setting as in ooh323.conf [abcd] type=user context=default ip=62.193.1XX.2XX disallow=all allow=gsm allow=ulaw this gateway can make call, even if these lines are commented out and you restart the

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting

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 all, i want to have ooh323 connection between asterisk and cisco. in my scenario, asterisk is gateway and cisco is gatekeeper. this is my ooh323.conf file: [general] port=1720 bindaddr=192.168.0.227 gateway=yes faststart=yes h245tunneling=yes h323id=gw10 at test.com settracelevel=10 gatekeeper=192.168.0.212 context=from-trunk disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833

Hi to all, I was looking in google and also in this mailing list, but I dont find the solution to my problem, so I subscribe me to the list in order to post this e-mail and find the solution. This is the scenario: GSM Phone ----- GSM Network ---- TDM2406E --- ASterisk 1.4.0 (*) -------- VoIP Provider ------- Sip Phone or H323 Phone The problem is that I am generating calls from SIP and also

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a