similar to: ooh323 and h323, it accept the call even not added in h323.conf

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

Has anyone tried using ooh323 in Asterisk to talk H.323 to an Avaya S8500 running Communications Manager 4 software? I have a potential customer who has such a system, and wants an Asterisk box to talk to it. Apparently they don't have SIP installed. I've successfully got ooh323 talking between two Asterisk boxes, so am just after some confidence regarding the Avaya, or any gotchas to be

I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. Firstly, despite the warnings in h323/README, I decided to try using the distro-specific versions of openh323 and pwlib. Of course, the Makefiles in channels and channels/h323 assume that openh323 and pwlib have been specially compiled in $HOME, so I modified the Makefiles to look for headers and libraries in

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

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

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

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

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

anybody successfully running asterisk-callmanager scenario with h323 trunk (ooh323 channel driver in asterisk)? I'm using 1.2.12.1 & ooh323 from 1.2.4 add-ons, but seems, that ooh323 is ignoring dtmf digits from callmanager h323 trunk setup with chan_h323 is working fine with dtmf I tried all possible modes with ooh323, but without success, with chan_h323, I'm using default (rfc2833)

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

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

06.05.2015 10:58, s m ?????: Hello! I'm not h323 expert, may be somebody else can understand from this log what is happening, but I can't :-( Could you, please, provide log with tracelevel=6 in ooh323.conf ? Thank you! > 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

hello thanks Dmitry for your useful hints. i enable debug and solve my problem:). it was codec compatibility problem. but it is so strange; if i set codec g711alaw in cisco router and asterisk, i have the mentioned problem but if i set codec to transparent in cisco router, every thing will be ok. is there any difference between g711 codecs which cisco and asterisk utilize? On Wed, May 6, 2015

06.05.2015 10:06, s m ?????: > hello every body, > > i have big problem to configure h323 trunk between cisco router and > asterisk 11.13.1 which uses ooh323 module. any body knows if ooh323 > module can work with cisco routers or not???? (in gateway mode, it is > ok and register in cisco gatekeeper but i can not configure trunk h323) > > we use chan_ooh323 with cisco

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

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

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

Just a warning to people trying to use ooh323 with Asterisk 1.8.7. The example config file that comes with asterisk is called chan_ooh323.conf when it actually should be named ooh323.conf for it to work. Sent me into a panic when I was trying to install an H323 link to an Avaya server and the ooh323 module would not load because it could not find its configuration file. The file needs to be

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,