similar to: how to identify user using chan_H323

I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk. Has anyone gotten a Webswitch running? When I try to connect asterisk thinks everything works fine, while the webswitch just rings. I belive chan_h323 is picking the wrong port to talk at the webswitch on, however I'm not sure, nor am I sure how to fix it. Any clues/hints? A tcpdump is attached to show the session.

Hi all, can somebody explain this ? Thanks, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger Potsdamer Str. 18 A 14513 Teltow FON: +49 (0) 3328 3077731 FAX: +49 (0) 3328 334779 Email: thomas.haeger@beronet.com *******************************************

Hi all, i have tried to connect to a clarent gatekeeper. I have used both of h323 drivers chan_h323.so and chan_oh323.so. But no one can register to this gatekeeper. Our ip is activated on this gatekeeper. Maybe, i do wrong anything.... I have only set the "gatekeeper" option in the h323.conf or oh323.conf to the ip address from the gatekeeper. gatekeeper=x.x.x.x But no one of the

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

Hello All, Finally I realised that the Asterisk demo setup didn't include support for h323. (Maybe it should have been obvious) so I went to work out how to get the h323 channel running. I had openh323 and pwlib installed as I'd been playing with vocal so it didn't take long to do cd asterisk/channels/h323; make; make install; make samples, copy the pwlib and h323 libraries to

Hi all, i have read in the * whitepaper the following: "s: The "start" extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the "s" extension." I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong

Hi all, have anybody an idea where to get adsi phones in europe ? Thanks, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger Potsdamer Str. 18 A 14513 Teltow FON: +49 (0) 3328 3077731 FAX: +49 (0) 3328 334779 Email: thomas.haeger@beronet.com *******************************************

Hi all, i made a test scenario with two windoze machines: On the first one callgen323 is running in listening mode On the second one, callgen323 strarting 25 calls to the asterisk pbx, and the asterisk calls the first windoze machine. But after the second one make a few calls (mostly after 11 - 14) asterisk crashes with the only message : Segmentation fault. Are this to many calls for oh323

Hello! Tried to compile, but : [CC] chan_h323.c -> chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing pointer to incomplete type chan_h323.c:350: error: dereferencing pointer to incomplete type chan_h323.c: In function 'oh323_rtp_read': chan_h323.c:790: error: dereferencing pointer to incomplete type chan_h323.c:791: error:

Hello all, I got the following error compiling h323 support in the latest cvs. Below the error is a diff to the file that I got to make it work. I took an example out of sip as far as the syntax for ast_rtp_new. Not sure if it is correct or not, but it seems to work. Please correct me if I am wrong in the additional 2 arguements. Regards, Scott cc -g -pg -c -o chan_h323.o -march=i686

Hi All, I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris 10. When I compile asterisk, the build fails at chan_h323 with: ---------------------------------------------------------------------- chan_h323.c: In function `reload_config': chan_h323.c:2863: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_h323.c:2863: error: (Each undeclared

Hello List, I've been trying to compile Asterisk with H.323 support and, after correctly installing PTLib and H323plus (OpenH323), the Asterisk configure script still doesn't detect the dependencies as installed. I know they are correctly installed because after going into "[asterisk-source-directory]/channels/h323" and issuing a 'make opt', it correctly builds

Hi, We have been running asterisk in our telco interconnect box with ss7 and H323 configured. Everything ran find till now, however, today, it started crashing with the following messages: [Jul 28 14:56:55] WARNING[2968]: acl.c:541 ast_ouraddrfor: Cannot create socket [Jul 28 14:56:55] ERROR[2968]: chan_h323.c:962 __oh323_rtp_create: Unable to locate local IP address for RTP stream [Jul 28

Hi all, i use the PGSQL App, and i have found out, if you use a "(" or a ")" in your query the query crashes... My sample query was : SELECT count(*) from tbltest where fldtest='xxx' can somebody fix this ...?? Regards, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger Potsdamer Str. 18 A 14513 Teltow FON:

Hi all, i 've written a little Callgen script for generating calls through the outgoing spool directory. The calls goes over 8 ttyI devices to another pbx and come in through other 8 ttyI devices. But when i generate the calls, sometimes * register the calls but never initiate them. Especially when the files come to fast into the outgoing dir. What can be wrong ? Is it possible that the

Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas.

Hi all (Michael), how it is possible to get the ip address of the calling party ? (i know by using h323... but there're a few unknown segfaults...) and so i want to use oh323, but i have to get the ip from the caller to permit or deny the call with AGI. Is it possible at all ? Thanks, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger

What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp, ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was not created so no h323 support in asterisk. Just wondering when to expect it again because I was stupid and didn't make a backup of the asterisk code before wiping the directory for a rebuild. -------------- next part -------------- An HTML attachment

When the support for h323plus was announced for Asterisk 1.4.25, I tried to build this support in Asterisk. For this, I checked out the h323plus CVS from SourceForge, which reported version 1.20.beta5, and also the ptlib-2.4.2 source RPM from Fedora 10. I finally managed to build a chan_h323 for Asterisk 1.4.25, which apparently loads correctly, but now I see that I get a segfault whenever I

Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make