similar to: h323 with NAT

Hi everyone, Installed asterisk from yum repository but I think H.323 is not supported as I tried commands like this and they don't work: - *h.323 debug*: Enable chan_h323 debug - *h.323 gk cycle*: Manually re-register with the Gatekeper - *h.323 hangup*: Manually try to hang up a call - *h.323 no debug*: Disable chan_h323 debug - *h.323 no trace*: Disable H.323 Stack Tracing

I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4) The pwlib + opal packages don't satisfy Asterisk's configure script (to let H323 compile), so I removed those and added the latest ptlib + h323plus (from h323plus.org) I can compile ptlib and h323, but when I load chan_h323 in asterisk I get a segfault. I had to point LD_LIBRARY_PATH to /usr/local/lib with the

hello all, I have installed centos 5.5 ( linux text) and I have updated it with # yum install bison bison-devel================?ok # yum install ncurses ncurses-devel==========?ok # yum install zlib zlib-devel===============?ok # yum install openssl openssl-deve=======?ok # yum install gnutls-devel============ ==?ok # yum install gcc gcc-c++============?ok # yum install newt

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

Hi All; To enable the compilation for the addon that is coming with Asterisk 1.8 when doing compilation for the Asterisk, what should I do? Regards Bilal

Hi all OK this may sound like a good one but maybe someone can tell me. Simple context is - I want to unplug my existing conventional PBX from the Telco and place * with it's TE410P in between. Now the difficult part, the existing connection is E1 PRI (Q.931) with 6 B-channels. I need to be able to trigger a D-channel to the old PBX and a D-Channel to the Telco (Not BT!). Next I can put

Hello List, A few days ago I installed ViciDial on a server, and while looking to the default 'extensions.conf' file, I saw this line: exten => _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT) Can someone point me out to the Asterisk documentation part where explains how to use server IP's as extension number? I could not see it in the ATFOT2 book, and I would

I've been trying to build h323plus (both the release and svn) for chan_h323 on Fedora 10. No joy. I posted on the h323plus ml, but no response. Anybody here actually built it on Fedora? Wanna share your secrets, or even better a specfile? sean

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a

Hi All, First time caller - tried WINE today for the very first time, on OpenSolaris no less. Good news - was shocked and stunned how easy it was to get an old copy of Unreal up and running - everything (almost...) "just worked" with practically no configuration, got lovely smooth rendering using my crappy built-in Intel X3100 adaptor. However, I cannot get any audio output

Hi all, Ok, I've been bashing my head for a few hours now on this, trying to figure out if I've done something wrong, but everything seems to me hunky-dory. So here's the deal: 1. I've compiled the spandsp 0.0.2pre10 source code successfully and also the asterisk application associated with it. 2. Receiving a fax at asterisk works fine, at least appears to be working

Hi can anyone help me with this g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I../../include -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include

dear rsyncists, rsync is superb! but did i actually find a bug? i did: $ rsync -vRHSPa rsync://rsync.mirror.ac.uk/updates.redhat.com/7.3/en/os/SRPMS . into an empty local directory the result was 140 bogus hardlinks of the same file: -rw-rw-r-- 140 g g 6147582 2002/12/04 16:55:24 Canna-3.5b2-62.7.3.src.rpm -rw-rw-r-- 140 g g 6147582 2002/12/04 16:55:24

Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

The following occurs with code from yesterday's cvs (asterisk) and current OpenH323 code: [root@raid-2 h323]# make clean install rm -f *.o *.so core.* cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations - DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIA N -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING

Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But "core show applications" doesnt show me any "fax applications" and when I try to receive a fax:

Hi Forum, I am new and learning this research tool. was trying to run some tests on a censored dataset using the NADA library. Everything is hunky dory except 'censummary' throws an error and I am trying to figure out a reason for this. Please help! below you will see the error statement... thnx --nk > library(NADA) Loading required package: survival Loading required package: splines

Hi, anybody knows what's missing or problem why i cant compile asterisk-oh323 in my machine? i got this compiled successfully ...Openh323 - v1.12.2 ...pwlib - v1.5.2 except ...asterisk-oh323 - v0.6.5 here's the output as i run make... mkoy@sambag:~/voip/asterisk-oh323-0.6.5$ make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory

Hi all I am trying to compile Asterisk on RH9 I am have installed pwlib_1.5.2 readline-4.3-5 readline-devel-4.3-5 openssl-devel-0.9.7a-2 openssl-0.9.7a-2 openh323_1.12.2 I read that it has to do with pwlib not being installed correctly so i made sure it was in /usr/local correctly When i try to compile i get the follow errors.... I was wondering can anyone shed in light on this issue Thanks