similar to: Open channels

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 --

I have running Asterisk 1.2.5 with addons 1.2.1. on Fedora Core 4. I have installed ooh323 from 1.2.1 addons. How to upgrade addons to 1.2.2 version and install new ooh323 driver? Do I need to install addons 1.2.2 if I only need new ooh323 driver? Can I just untar addons, and run "make clean; make; make install" and then execute following cd asterisk-ooh323c ./configure make make

Does ooh323 from asterisk-addons 1.2.1 support alaw codec? This is what is written in h323.conf.sample that can be found in asterisk-addons dir. The codecs to be used for all clients.Only ulaw and gsm supported as of now. Default - ulaw ONLY ulaw, gsm, g729 and g7231 supported as of now disallow=all allow=gsm allow=ulaw So, it shouldn't support alaw, but I manage to establish calls with

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.

Situation. I call out from SIP phone over h323 trunk and called person decides not to pick up (on mobile phone they press red button - NO - hang-up). Until the called person press the NO button, I can hear ringing. When called person press the button, I don't hear anything. Asterisk waits until timeout and than ends the call. How can I get busy or some other appropriate signal on SIP phone

How should I upgrade addons form ver. 1.2.1. to 1.2.2.? I'm particularly interested how to upgrade ooh323 channel driver. -- Tomislav Parcina tparcina#lama.hr

Hi group! I'm having problems with ooh323 (ver 0.3?!? - the one that comes with asterisk addons 1.2.1) and I need to know what h323 channel driver you use in production? Have a nice day! -- Tomislav Parcina tparcina#lama.hr

In article <1251.165.146.69.140.1144596935.squirrel@www.ecntelecoms.com>, yusuf@ecntelecoms.com says... > Hi, > > I have had the exact same problem last week. I have not yet solved it. > So instead I am using ooh323, but would prefer to use oh323. Can anyone > help? I'm glad that I'm not the only one :)) Hopefully we'll find solution to this problem. --

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 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

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

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

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

Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")

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 all. Trying to setup H.323 via Asterisk between a PLANET H.323 box and my SIP phones. When calling from the SIP phones, it connects but quickly disconnects citing the following error message: **** --- build_peer +++ build_peer +++ reload_config +++ ooh323_do_reload -- Executing Dial("SIP/yyy-2965", "OOH323/203@xxx") in new stack --- ooh323_request - data

Hello, I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have done this before between Asterisk 1.6 and Avaya but had some issues placing external calls from the Asterisk to the Public network which is connected to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is outdated and has no support. On the Asterisk side I have Aastra 6731i SIP phones

Dear All, It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a

Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0 at fromtransfer:1] NoOp("OOH323/7272-6385", "") in new stack -- Executing [0 at fromtransfer:1]

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