similar to: Chan_h323 one way audio

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,

Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can "hear" me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external

Hi all, I try to ask helps again about chan_h323 extensions. I define this in h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 allow=gsm allow=ulaw gatekeeper = DISABLE context=default [gm1] type=friend host=192.168.1.20 context=default [gm2] type=friend host=192.168.1.25 context=default and I have in extensions.conf : [demo]

I have built Asterisk on SuSe Linux 7.3 on an Ultra 2 Sparc WorkStation. I am listing the modification I had to do for the benefit of anybody else who wants to use Asterisk This workstation is equipped with one 400 MHz RISC UltraSparc II CPU, 256 MB RAM, Two 9 GB 10,000 RPM UltraSCSI Disks. I have a gatekeeper running on this machine, I had to do the following modification to build * on Sparc:

Hi, Has anybody managed to get callerid properly set on a call from local to asterisk SIP endpoint through h323-pstn gateway to a regular phone. I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it. When I place a call to pstn I'm not receiving 12125551234 as the clid, but a number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other

Hi all: I've got some core dumps when I use chan_h323. I dial an extension using h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes * hangs, sometimes not. The client used for test es SjPhone (http://www.sjlabs.com/). This is the data for one core dump: (gdb) bt #0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790 #1 0x41f8879c in create_connection

Hi, I'm wondering if there are any plans on adding secondary gatekeeper support to asterisk h323 channel drivers. Also I've noticed that chan_h323 is crashing asterisk at startup if primary gatekeeper is not available. Wouldn't it be a more correct behavior if it doesn't crashing but continue registration attempts in the background? Didn't test it with chan_oh323. Thank you.

i have tried to install oh323 but it has failed to load this module please help [chan_oh323.so]WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/local/lib/libh323_linux_x86_r.so.1: undefined symbol: _ZN13PASN_Sequence17PreambleDecodeXERER11PXER_Stream WARNING[1024]: File loader.c, Line 368 (load_modules): Loading module chan_oh323.so failed!

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

Hi, Someone released a succefull interconnection in H323 with WTL equipement ? I'm trying to do that with an IPNx. But get dead air. With chan_oh323 it's fine, all works. With chan_h323 => dead air. The configuration is GW to GW. This is my configuration from h323.conf: [general] port=1720 bindaddr=my.ipaddr dtmfmode=rfc2833

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten => 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work).... Thanx in advance, Victor...

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Hi, I also had some problems using chan_oh323 together with gnugk. * <-> gnugk <-> h323-phone When I called the phone and hang up, befor the phone was picked up, the h323-phone continued ringing. The same, when the h323- and some sip-phones were called, and the sip-phone picked up the call first. (It is annoying, when you are talking to someone at the phone and the phone on the

Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm

I am trying to figure out the correct way to send my E.164 ID with chan_h323 and or chan_oh323 as my H323 provider requires this in the format of 'account-pin'. With chan_oh323 I have been able to register with the gatekeeper and can recieve incomming calls, but outgoing calls do not work. With chan_h323, I can call H323 clients (netmeeting, ATAs etc) but cannot place a call through my

---------- Forwarded message ---------- From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> Date: Oct 4, 2007 12:56 PM Subject: Re: [asterisk-dev] chan_h323 and chan_oh323 compatibilities To: asterisk-dev at lists.digium.com Hi On Thu, Oct 04, 2007 at 11:46:30AM -0300, Caciano Machado wrote: > I'm receiving a lot of warning messages from my Asterisk > 1.2.5/chan_oh323 every time

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

I have no new information, just a note of encouragement to those traversing the bowels of h323: I've been trying to get h323 working with asterisk for several months now, trying with chan_h323 & chan_oh323 with all kinds of different combinations. As with several folk on the list, I've had no luck. Either I had no audio, or I could only receive calls, or I could dial but no had

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