search for: ooh323

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 OOH323/(null)-b7db8aa0 at internal,s,1 still failed so falling back to context 'default' -- Executing [s at default:1] Playback("OOH323/(null)-b7db8aa0", "vm-goodbye") in n...

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 t...

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...

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 have to do to capture...

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

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("OO...

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....

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...

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 ooh...

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] NoOp("SIP/6052-00000ab6", "") in new stack -- Executing [0 at fromtran...

...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 the h323.conf? i want asterisk and gnugk in the same machine (lets say ip 192.168.0.10). then sip_client at 192.168.0.100 (telephone number assigned for instance 100) and H323_client at 192.168.0.200 (telephone number assigned 200) how do i confi...

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 203@xxx format 0x4 (ulaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- 203@xxx --- onNewCallCreated ooh323c_o_22 --- find_call +++ find_ca...

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...

...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 what I can tell, the cause is "condition 20 on ooh323". Any suggestions as to the cause? http://www.elastix.org/component/option,com_fireboard/Itemid,55/func,view/catid,3/id,41480/lang,en/#42715 Dec 29 10:25:01 VERBOSE [15027] logger.c: -- Remote UNIX connection Dec 29 10:25:01 VERBOSE [31438] logger.c: -- Remote UNIX connection discon...

Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", "SIP/6166 at asterisk") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/6166 at asterisk > 0x7fa9d4007660 -- Probation passed - setting RTP source address to 10.18.0.19:26052 -- SIP/asterisk-0000000c is m...

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...

Hi all, I have installed Asterisk-addons-1.4.5. I was getting error cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory So, I did following steps: cp asterisk-ooh323c/.libs/libchan_h323.1.0.1 asterisk-ooh323c/.libs/libchan_h323.so.1.0.1 make install make samples It worked properly.But still I am not getting ooh323.conf in /etc/asterisk Please help me. Am I doing something wrong? What I should do to get ooh323.conf Thanking you, Preeta Pandey Please do not...

Hi! I try to get rid of some part of CALLERID(name) but I cant realy figure out a way to do it. For example: CALLERID(name) = "Martela (fax)" I am just looking for the part before ? (? in my case ?Martela?. I can?t serch for ? ?, could be many ? ?, but only one ? (?, thought i could do something like: exten => 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1}) But that gave me

...n 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 ooh323 debug(in asterisk11.13.1 system): ooh323_get_rtp_peer OOH323/peer-2-5 -> (null):0, 1 this ia h322_log: 10:42:10:835 Processing MakeCall command ooh323c_o_3 10:42:10:835 Created a new call (outgoing, ooh323c_o_3) 10:42:10:835 Enabled RTP/CISCO DTMF capability for (outgoing, ooh323c_o_3) 10...

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 to me tha...