similar to: OH323, RxFax and codecs

Dear All, I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but it don't work. oh323 driver don't want connect to gateway with g729, it's work if I only use in oh323.conf one codec ( g729 ). If I enable 2 or more codecs - always in use other codec: -- Executing SetVar("IAX2[4010@4010]/1", "OH323_OUTCODEC=g729a") in new stack -- Executing

Hello, I am facing some problems with OH323 1. I am using exten => 223,1,SetVar(OH323_OUTCODEC=gsm) or g729 exten => 223,2,Dial(OH323/x.x.x.x) but the call always go out in aLaw . 2. With OH323 I find one way audio . 3. How can I use ASTCC with OH323 , I have tried to use dialstrings like exten => 223,2,Dial(OH323/192.168.0.1/1234) to send the dialed number as 1234 , but it does

I've complied the OH323 .so successfully and can easily receive calls from my H323 gatekeeper (using 711u), however it seems that all outgoing calls are refused and I'm getting "reason 23 (Temporary failure)" as an error code which I can't find documented everywhere. My H323 gatekeeper needs a 001NXXNXXXXXX to dial out to the PSTN even if I'm in north america (Montreal)

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

Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1). my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not

This looks like the same case as the thread: Problem ATA-711-723-Oh323-Asterisk that was posted back in August. Although, given the nature of trampled memory problems it could be almost anything. The Oh323 library dies in a call to realloc. This is almost certainly indicative of someone stomping on memory somewhere previously, or else a memory bug in the pwlib library itself. (Which I find

Hi, I have a communication partner who needs G.729. When I disable in oh323.conf all codecs except G.729A, everthing is fine, except that I can't receive calls with other codecs from other partners via H.323. That's why I enabled also GSM and G.711 in oh323.conf and put a SetVar(OH323_OUTCODEC=g729) in my extensions.conf for that partner. Now I see in the console log: -- Executing

Hello I had this codec negotiation with oh323 call. i used G723 codec and the provider had G729 as first priority. In this situation what ever number i dial i used get "No one there to answer the call". As soon as i changed my codec to G729 the call went through but had other problems, which i got away by dowloading the latest code for oh323. Has anyone seen this problem? or it is

Hello! I have been doing some tests with call transfers and I have been looking at the CDRs that Asterisk generates. Scenario 1: A calls B B answers and does a blind transfer to C (using # key) C answers and talks with A Scenario 2: A calls B B answers and does an attended transfer do C (using the phone's transfer key) C answers, B hangs up, and C talks with A For scenario 1, the CDR shows

Joao, Thanks for sending the Installation tips as pasted below. It works. Seshu ---------- Get oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz Get pwlib from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz Get asterisk-oh323 from

Hi!, I tride comile oh323. I have downloaded local versions of pwlib oh323 (both Janus patched). I get following errors on asterisk-oh323-0.6.5: for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/home/death/1/asterisk-oh323-0.6.5/wrapper' ./check_ver /root/pwlib pwlib ./check_ver /root/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o

after editing the Makefile according to my directories ive compiled asterisk-oh323-0.6.5 but i got these errors can any body help me with this : [root@terra@net asterisk-oh323-0.6.5]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/home/wassim/asterisk-oh323-0.6.5/wrapper' ./check_ver /home/wassim/pwlib pwlib ./check_ver

Hi, well, I really need your help here. I have tried compiling oh323 many times and I always get the following error when trying to "make opt" open h323. Any ideas?! Compilation Error: -------------------------- g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib ./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lexpat

I'm and Asterisk@home user - been so now for almost a year. Lately, I've upgraded to the latest & greatest.. (which is built on 1.2.5) and am unable to install oh323. I've already asked over at the (A@H) Sourceforge forum but no one seems to think it worth answering. The error I get is pretty obvious but I don't know where to go from here. More importantly, I need to have

I'm following the instruction from Jo?o Amaro from the page http://lists.digium.com/pipermail/asterisk-users/2005-February/090752.html Everything went fine until I run the 'make' command under asterisk-oh323-0.6.5. I got the error message chan_oh323.c:5220: too many arguments to function `ast_channel_register' I have attached the error message. I'm running asterisk CVS

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

Hi, I'm using OH323, mostly with success, to interface Asterisk to a provider's switch (World Telecom INX). I have noticed a particular effect, and I wonder whether anyone else has seen the same? The effect is audio flutter (almost like the flutter one gets on MF or HF radio sometimes) which only happens intermittently. Audio coming into Asterisk is unaffected, as proved by using the

Hi, I have downloaded files and also local versions of pwlib oh323 (both Janus patched). Both libraries compile fine, but I get following errors on asterisk-oh323-0.6.5. Readme is a bit confusing since it doesn't mention which local libraries should be downloaded from inaccess to get everything working OK. I've also tried with/without patching oh323 with supplied patch. Any hint, advice

Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be quite disturbing for the users. Any help would be very welcome. thank you. Yves

I am trying to compile oh323 and having the following error. Can anyone help please?! This is my third post. These are the versions I am using: Compilation Error: -------------------------- g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib ./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lexpat -lresolv -ldl