similar to: H.323 trunking

Hi all, this is my actuel setup [SIP 2005]--[asterisk]--H.323 Trunk--[PBX]--[ext. 8900] Linux CentOS 3.3 (2.4.21-20.EL.c0) asterisk-1.0.1 asterisk-oh323-0.6.3b openh323_1.12.2 pwlib_1.5.2 Calling from SIPphone to the extension 8900 works always. Calling from 8900 to SIPphone works only sporadicly without any recognizeable pattern. Find below the output of the debug command: asterisk

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

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

Hello.... Asterisk is compiled and running perfectly... But when i try to compile Channel_h323... that's another story :-|. I'm compiling using RH9, OpenH323_1.12.2, pwlib_1.5.2, i compiled and installed them myself... GnuGk is running smoothly using them. Also my intention is to terminate calls from a SIP client through Asterisk and into my GnuGK... has anyone accomplished this ?

Hi, I'm trying to get h323 communication working between asterisk (0.7.1) and Innovaphone Gatekeeper + innovaphone phones. chan_323 installed OK with currently recommended pwlib_1.5.2 and openh323_1.12.2. Registration asterisk with the gatekeeper works OK, externsion for my test(sip) phone gets registered with gatekeeper. when establishing a call between a h323 phone and asterisk I run into

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

Hi, I have some data (lots of year,distance pairs), which I can straightforwardly boxplot: dists <- read.table("movedists.dat") with(dists,plot(as.factor(V1),V2)) If I want to plot these data as violin plots using vioplot, this approach doesn't work: > with(dists,vioplot(as.factor(V1),V2)) Error in Summary.factor(c(1L, 1L, 1L, 1L, 1L, 1L, 1L, 1L, 1L, 1L, 1L, : min not

hello: i want to test the g729 with asterisk. my scenario is sipp(ulaw)->asterisk1 with g729->asterisk2 with g729. I want to test g729 module with asterisk-1.4.21, when i make calls from asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip in asterisk 1 is with codec g729 and enforce that use g729, the sip in asterisk 2 also work with G729 only, but asterisk 2

Hi there ! I searched the whole web to find some helping information about H323 Control Protocol, but there is no way to find that information. We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2 + 'asterisk-oh323_1.5 channel driver + wrapper' and configured the dialplan for using our H323 Endpoints which are ip200 Innovaphones. Besides, we also use Gnomemeeting but

After much reading and fiddling - I have the gnugk GateKeeper running and can make calls from the H323 phone to the sip phone. Voice works bi-directionally.. Calling from SIP to H323 gives me a problem... Both gnuGK and Asterisk are on the same box. Someone said this was OK. Others said No. I added a second IP (eth0:1) and told gnuGK that was HOME. How do I lock asterisk to the other (eth0) IP -

I need some help with getting the following to work SipPhone <------> Asterisk <------> H323 GK (quintum) And H323Phone <------> Asterisk <------> H323 GK (quintum) I have tried to run the Asterisk from the newest ports and could after some digging around in the configs register the SipPone to Asterisk and Asterisk to the H323 GK. But when I try to make a call from

Hi all, I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the softclient. Any ideas? Thx! sip.conf [2005] type=friend callerid="OptiPoint" <2005>

Hi there ! I searched the whole web to find some helping information about H323 Control Protocol, but there is no way to find that information. We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2 + 'asterisk-oh323_1.5 channel driver + wrapper' and configured the dialplan for using our H323 Endpoints which are ip200 Innovaphones. Besides, we also use Gnomemeeting but

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

This series adds new APIs to support btrfstune. Chen Hanxiao (2): New API: btrfs-image New API: btrfs_image_restore daemon/btrfs.c | 87 ++++++++++++++++++++++++++++++++++++++++++++++++++++ generator/actions.ml | 45 +++++++++++++++++++++++++++ src/MAX_PROC_NR | 2 +- 3 files changed, 133 insertions(+), 1 deletion(-) -- 2.1.0

Hello, I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 installed but failed. I applied the patch to the required OpenH323 library according to the instructions, and set the proper directories in the Makefile. Here is what I receive after I issue make: ******************************* g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

Hello I?ve just install last cvs version (Mon Sep 6) of Asterisk with asterisk-oh323-0.6.3b and pwlib-v1_6_6-src.tar.gz, openh323-v1_13_5-src.tar.gz and . this is the error loading asterisk with chan_oh323 module:: [cdr_csv.so] => (Comma Separated Values CDR Backend) [cdr_manager.so] => (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found

Dear R-help, I have a data set consisting of measurements made on multiple subjects. Measurement sessions are repeated for each subject on multiple dates. Not all subjects have the same number of sessions. To create a factor that represents the session, I do the following: data <- read.csv('test-data.csv') # data appended below data$date <- as.Date(data$date,