similar to: Setting the H323 Callerid sent by asterisk (using chan_h323)

I have one and only one reason to use OH323, and that is it works with Cisco's CallManager. The standard H323 channel has one-way audio issues when connected to CCM and fixing them has been identified as a low priority/never going to happen task. Dan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

The hack below is for OpenH323, not Asterisk. This is not an Asterisk problem AFAICT. I am posting it here so that any other Asterisk user with a similar problem might benefit from it. I may or may not post it to an OpenH323 list, but since both variants of the H.323 channel in Asterisk use non-current OpenH323 versions, it may not be of any benefit to anyone anytime soon if I went that route!

Hi I have found in http://lists.digium.com/pipermail/asterisk-users/2004-July/056111.html (Hack to make * -> (H323) -> CCM -> IOS GW work) that i need a special version of chan_h323, because of the External RTP problem. Do you know exactly which version is it? Or do i need an unofficial patch? Thanx Andr?s

Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =

>Date: Thu, 06 Apr 2006 12:47:00 -0700 >From: Jeremy McNamara <jj@nufone.net> >Subject: Re: [Asterisk-Users] Pickup() h323 >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <44357034.5090802@nufone.net> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Dan Austin wrote: >

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

Greetings- As most of you who monitor this list know, I've been messing about with Asterisk -HEAD, Cisco Callmanager, and the nufone H323 channel driver here for some time- with pretty decent success. I'm hoping to cash in a chip here- I've run into something that is probably a very simple answer, yet not found a decent reference to resolve it. Scenario- -HEAD as of last week

Some of you may remember back in May the thread on using Asterisk as a voicemail server for a Cisco Callmanager system. My own Callmanager system is integrated into an Asterisk server for voicemail (and other things). Back in May I was using H323 for integration, but since I've upgraded to CCM 4.1 I have switched over to SIP. The integration with H323 required using Call forwarding to send

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

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

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, i'm searching and trying, but can't get it working. I'm trying to send calls from Cisco Callmanager to Asterisk with oh323 channel driver. Therefor the asterisk is defined as a H323 Gateway in the Cisco Callmanager. The Call comes from CCM to Asterisk and it works but i didn't get the called number. This is needed because i want to make Voicemailboxes. If i connect via

We have an odd problem, where inbound H323 (chan_h323) calls will sometimes cause a Ringing Congestion that appears to keep the channels open and never release it until we kill and restart asterisk. These "Ringing Congestions" start to pile up, which eventually crashes Asterisk. H323 Gateway -> Asterisk (chan_h323) -> Tor2/PRI -> PSTN Has anyone ran into this problem or

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 am using Nu Fone Network's h323 drivers. I can place H323 calls using following in extensions.conf file, exten => _1732.,1,Dial(H323/${EXTEN}@192.168.1.2) If I need to use h323.conf to do the same I cannot configure h323 to do the same. I get everyone is busy message and I do not see IP packets being generated by * trying to communicate to 192.168.1.2. Can someone point out what I

I'm attempting to get chan_h323 working on Asterisk CVS-STABLE. I've compiled it ok using the Janus release of pwlib/openh323, by editing the makefile as per the comments. Call setup and cleardown seems to work fine, but no audio is being passed in either direction. Doing an "h.323 trace 9", I noticed the following sequence at the end of the call setup: h323.cxx(1685)

I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration

anybody successfully running asterisk-callmanager scenario with h323 trunk (ooh323 channel driver in asterisk)? I'm using 1.2.12.1 & ooh323 from 1.2.4 add-ons, but seems, that ooh323 is ignoring dtmf digits from callmanager h323 trunk setup with chan_h323 is working fine with dtmf I tried all possible modes with ooh323, but without success, with chan_h323, I'm using default (rfc2833)

Heya, I changed the Makefile of the h323-channel-makefile (I downloaded cvs of a couple of hours ago) so that I don't have to rebuild pwlib and openh323, but can use the precompiled versions. I'm using pwlib 1.8.3 and openh323 1.15.2. There aren't many changes. I replaced OPENH323DIR with OPENH323INC ,which points to /usr/include/openh323 for me and OPENH323LIB, which points to

Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h323.conf like: [office] type=h323 prefix=9 context=outbound I get a message saying: