similar to: ooh323_indicate: Don't know how to indicate condition 20

Hi, I recently changed my version of asterisk to 11.XX, and I see a waning with h323, with version 1.8 did not have these warning I have connected one avaya ip office 500 h323 with asterisk and the 1.8 version did not have these messages Oct 23 17:20:35] WARNING[7593][C-000000aa]: chan_ooh323.c:1413 ooh323_indicate: Don't know how to indicate condition 33 on ooh323c_60 [Oct 23 17:20:35]

Everyone, I have been trying to connect a PBX with H323 IP trunks with g711 codec to my Asterisk server running ooh323 service. I can place calls to and from either the Asterisk, or PBX with no problem, but when I try to pickup the call on either end, the phone hangs up immediately. Debug shows normal to me but at the last few lines of data there is an error shown that I have not been able to

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

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

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

Hi, I am using asterisk 1.4. I have confgured the musiconhold.conf file. However, when i make a call and then hold the call it does nothing. in the CLI i do not see the starting/stopping musiconhold messages. i am making calls from sip to h323 using asterisk assip/h323 gateway (with gnugk and ooh323). i get the following messages when putting the call on hold: -- Executing [204 at default:1]

Call-Bilal*CLI> module load chan_ooh323.so Loaded chan_ooh323.so [Jun 17 20:23:32] NOTICE[2392]: chan_ooh323.c:2506 reload_config: Unable to load config ooh323.conf, OOH323 disabled Loaded chan_ooh323.so => (Objective Systems H323 Channel) Again, from make menuselect, if I selected chan_ooh323 from the Add-ons and I selected ADDON from module embedding. Then I ran ./configure and make. I

Hello! I have a problem with build astersik-addons-1.4.7 on Solaris 10. When I tried to do "make" I got such error: * chan_ooh323.c: In function `reload_config': chan_ooh323.c:2053: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_ooh323.c:2053: error: (Each undeclared identifier is reported only once chan_ooh323.c:2053: error: for each function it appears

Hi all, I am a bit out with the Asterisk 1.4.4, after I complied and installed the Asterisk and I got such error messages [Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol:

Hi all, Recently I am trying to install the Asterisk 1.4, I has some error while loading the following modules, can some one help on those issues? Error during loading the modules; Basically, chan_ooh323.so, and res_config_mysql.so "" [Jun 23 12:10:01] WARNING[30257] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined

Hi, We want to upgrade both our servers to asterisk 1.8, the one from Romania and the one from Chicago, but for the moment I`m trying to install Asterisk 1.8 on a test machine running CentOS 5.5 with the kernel: Linux asterisk3 2.6.18-194.17.4.el5PAE #1 SMP Mon Oct 25 16:35:27 EDT 2010 i686 i686 i386 GNU/Linux . I`ve tried many things from the forums and mailing lists but none seemed to help me.

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, Does Asterisk_addons_1.4.2 cant be use in the new Asterisk release or no one really want to share their experiences? Since this project is belonging to everyone within this list, why still no one really want to share the experiences and to growth the Asterisk to the next level by keeping their secret in behind. As see, Asterisk 1.4 has so many feature improvements, and it's

The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

Hi all, Sorry for cross posting. I attached my chan_ooh323 patches (asterisk-addons-1.4.5) when codec negotiation patch changes applied to asterisk-1.4.17. Please let me know whether my patches are correct or not. thanks in advance, Ganbold -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dears; Actually, the needed file name to be ooh323.conf and not chan_ooh323.conf, so I copied the file from chan_ooh323.conf to a new file name ooh323.conf and it is working fine.

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can

I terminate some calls on a h323 device (a quescom gsmgateway) from asterisk 1.2.3 with ooh323, the customer is complayining about choppy sound on most of the calls, the only warning message I can see is : src/chan_h323.c:944 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_102 (the calls sounds perfectly with iax/zap termination and the quescom seems to work fine with

to have lpthread but with no luck , still doesn=E2=80=99t work. Thanks for the answers guys, Bogdan ------=_NextPart_000_0728_01CB7CD3.380311A0 Content-Type: text/html; charset="utf-8" Content-Transfer-Encoding: quoted-printable <HTML><HEAD></HEAD> <BODY dir=3Dltr> <DIV dir=3Dltr> <DIV style=3D"FONT-FAMILY: 'Calibri'; COLOR: #000000;