similar to: Is H323 supported when installing Asterisk from Digium Yum repository?

Displaying 20 results from an estimated 2000 matches similar to: "Is H323 supported when installing Asterisk from Digium Yum repository?"

2011 Apr 27
1
h323 with NAT
Hi list, I've been beating my head for about 3 days on this one. I have Asterisk 1.4.41 installed using openh323. As long as I'm inside my firewall, everything is hunky-dory. When I move to server on another subnet, I'm still able to connect, but no longer have sound. Any good pointers or suggestions? Thanks Danny Nicholas
2010 Jun 20
1
Compiling H323
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
2011 Jan 16
0
chan_h323 and menuselect dependencies problem
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
2004 Aug 04
5
H323 Call Dropping
Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call
2011 Jan 18
0
Asterisk SlackBuilds for Slackware Linux
Hello List, To whom it might concern: I have been working in some SlackBuilds (script for making Slackware Packages) for my personal use, but thought they might be useful for someone else here. Beside of the exceptional distributions used so far (CentOS, Debian, Ubuntu, etc.), you might want to test Asterisk on a Slackware Linux box, as it offers outstanding stability and flexibility as
2004 Jan 29
0
Register to h323 gk
Hello group, I am trying to register to a opengk h323 gatekeeper using chan_h323. The gatekeeper expects me to register a username like 31201234567@gatekeper.com with a password secret and an e164 of 31201234567. Thus I put the following in the config file: [general] gatekeeper=w.x.y.z. AllowGKRouted=yes [31201234567@gatekeer.com.com] type=h323 e164=31201234567 secret=geheim
2009 Apr 02
1
Anyone actually built h323plus on Fedora?
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
2009 Nov 30
2
No application 'ReceiveFAX'
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:
2024 Feb 10
2
Joining Windows 10 Domain Member to Samba AD/DC
On Sat Feb 10 15:31:47 2024 Mark Foley <mfoley at novatec-inc.com> wrote: > > On Sat, Feb 10, 2024 at 2:20?PM Mark Foley via samba > <samba at lists.samba.org> wrote: > > Does chrony have to be built in some special way to enable ntp-signd? > > Needs to be configured with "--enable-ntp-signd". I may have to build from sources. I downloaded from the
2008 Dec 04
1
OT - Is sourceforge OpenH323 active ?
Hi, A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Mar 13
1
RE: In Asterisk 1.4.x, Why Digium has two H323 channels?
Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more,
2018 Feb 19
2
Best Power FE700VA?
On 02/13/2018 03:49 AM, Charles Lepple wrote: > On Feb 12, 2018, at 11:53 PM, David Melik wrote: > > On 01/21/2018 05:53 AM, Charles Lepple wrote: >>> If you hook up a terminal emulator to the comms port (1200 baud, 8/N/1 unless stickers and/or configuration DIP switches indicate otherwise), and type "f" <return>, do you get a status line like one of the
2009 Jun 02
0
Segfault on unload of chan_h323 in asterisk-1.4.25
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
2004 Jan 23
1
PSTN incoming - both SIP & H323 always arrive in default context :-?
Some of you may remember seeing my issue using SIP for incoming calls from the PSTN: http://voip-info.org/wiki-Asterisk+cisco+FXO i.e. all incoming calls arrive in the default 'bogon-calls' context. Well, I tried again using H.323 & get exactly the same result (both for chan_h323 & chan_oh323) i.e. all attempts to put a type=peer in sip.conf or a type=user in h323.conf for
2003 Jun 16
2
h323 compile error
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
2011 Apr 20
2
issue with installtion asterisk
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
2005 Apr 10
2
Problems trying to compile H323 from CVS-STABLE
I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. Firstly, despite the warnings in h323/README, I decided to try using the distro-specific versions of openh323 and pwlib. Of course, the Makefiles in channels and channels/h323 assume that openh323 and pwlib have been specially compiled in $HOME, so I modified the Makefiles to look for headers and libraries in
2006 Apr 03
6
Pickup() h323
Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing
2005 Aug 10
1
h323 error when trying to start Asterisk
Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Aug 10 09:09:18 WARNING[7824]:
2010 Mar 24
6
Restarting Asterisk using a script - Thanks to all -
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