similar to: Can I install latest oH323 on *@home

I have successfully installed oH323, but when I make a call the recipient can hear me but I can't hear anything at my end, so voice is only travelling in one direction. Client Softphone (SIP) => Asterisk => H323 Gateway. All Codecs have been loaded. Any help would be appreciated. Ade. --------------------------------- To help you stay safe and secure online, we've

Hi all, with asterisk-1.2.6, asterisk-oh323-0.7.3, openh323-Mimas_patch2, pwlib-Mimas_patch2, on FC3 kernel 2.6.5-1.358, I can make some h323 calls with no problem. However, on certain numbers, asterisk just crashes on the console i get: -- H.323 call to 6199375800@196.x.x.x with codec(s) g729 -- Outbound H.323 call to destination '6199375800@x.x.x.x', channel

Hi; I use the following asterisk, openh323, pwlib: asterisk = cvs-head-03/09/05 openh323 = 1.13.5 pwlib = 1.6.6 asterisk-oh323= 0.7.1 Asterisk, openh323, pwlib were compiled successfully but when I try to compile Asterisk-Oh323-0.7.1 , I got the following error: chan_oh323.o chan_oh323.c chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory ......... ...........

i have tried to install oh323 but it has failed to load this module please help [chan_oh323.so]WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/local/lib/libh323_linux_x86_r.so.1: undefined symbol: _ZN13PASN_Sequence17PreambleDecodeXERER11PXER_Stream WARNING[1024]: File loader.c, Line 368 (load_modules): Loading module chan_oh323.so failed!

Hi group! Does DTMF inband work with oh323 channel driver ver. 0.6.7? If yes, how to enable it, make it work? I have tried with "inBandDTMF=yes" in general context of oh323.conf, but I get this message when I * is starting. [chan_oh323.so] => (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing

I'm running: OS: FreeBSD 6.0 Asterisk: 1.2.4 Installing OH323: 0.7.3 I have this error when compiling chan_oh323.c: In function `reload_config': chan_oh323.c:4677: warning: implicit declaration of function `sscanf' chan_oh323.c: At top level: chan_oh323.c:3244: warning: 'update_call_ids' defined but not used gcc -shared -Xlinker -x -g -o chan_oh323.so chan_oh323.o

Hi all, I got serious problem here, I hope I ask on the right place here (sorry if I am wrong). I have used asterisk 1.2.17 with openh323 ver. 0.7.3, for integrating between SIP Gateway and H323 Gateway, it runs about 6 months. But, recently I think it doesn't work anymore...I can't call from SIP Gateway to H323 Gateway. I try to debug oh323 by using : # oh323 debug toggle But I got no

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the

I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I

Hello all. Has anybody work with asterisk version 1.2.2 and oh323 module? When I try to make it show: wrapendpoint.cxx:800: error: expected primary-expression before ')' token wrapendpoint.cxx:800: error: `PIsDescendant' undeclared (first use this function) wrapendpoint.cxx:800: error: (Each undeclared identifier is reported only once for each function it appears in.)

Hi all, I'm trying to set up a test case for an ISP featuring an asterisk server and a couple of linksys rt31p2-na routers registering on it. Instead of using dsl lines, i'm trying to plug the * server and the routers on a cisco switch, just to test their functionality. I have created a vlan and a subnet on the switch and set up the ip addresses of the routers in that subnet. When i plug

Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday).

Hi all i have been succesfully using OpenH323 (oh323) for a few months. the versions are: asterisk CVS HEAD 19-07-2005, OpenH323 v1.13.5, PWlib v1.6.6, asterisk-oh323-0.7.2-pre1 I now have moved to Asterisk 1.2.4, so as per the directions i am using: Asterisk 1.2.4 pwlib_Mimas_patch2 openh323_Mimas_rc2 asterisk-oh323-0.7.3 The problem is that when asterisk starts it fails on loading the module

Hi, I'm using Asterisk 1.2.3 with the asterisk-oh323 channel driver, version 0.7.3. Pwlib is V1.8.7 an OpenH323 is V1.15.6. Following CallFlow: SIP-UA -> OpenSER -> * -> CCM OpenSER routes all calls with prefix 60 to Asterisk, where I've configured following extension: exten => _60.,1,Dial(OH323/${EXTEN:2}@v.w.x.y) v.w.x.y is a Cisco Callmanager where Asterisk is

Hello Im having problem building oh323 on SuSE Linux 9.3, I have build the openH323 and pwlib and Im getting the following error: g++ -Wall -felide-constructors -x c++ -Os -DP_USE_PRAGMA -ffunction-sections -fd ata-sections -D_REENTRANT -Wall -fPIC -I/usr/src/pwlib/include -DPTRACING -I/usr /src/openh323/include -DHAS_OSS -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSI ON=\"1.6.6\"

Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf

Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652&scid=29 -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie

I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, First off only been using Ubuntu for a week, so really very new to this, if I need to do anything in the terminal please be really basic! Running Ubuntu 10.10 I installed WINE no problem, version 1.2.2 Adobe reader has been installed via Synaptic Package manager V9.4 I then installed via WINE Adobe Digital Editions, this has also loaded and is running. I have bought some books and