similar to: one way audio h323 callmanager

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I've just finished my upgrade to asterisk RC2. I need to have H323 support, and in the last months i've been using the chan-oh323 with good results. My question is: anyone in the list have made tests with both chans (oh323 and h323), which is best ? For this installation i don't need the gatekeeper support, i just want to

* compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled --

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? <http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration> I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi

hi guys, i'm using sept 30 cvs and oh323 5.5 i'm having 5 second latecy(on only 1 audio path) when a call is transferred.... the scenario is this: sip--------->asterisk----->h323:operator (who then transfers the call) ---------------->h323:destination ------------------audio path 5-second latency---------------->

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 guys, i'm encountering one way audio on cvs using netmeeting and chan_h323.so is there a quick fix or workaround for this? compiled using openh323 1.12 pwlib 1.5 i also saw this in earlier version of openh323 and pwlib.... thanks for any info ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this

Hello all We succesfully added a H323 Gateway to our CallManager 3.0 that resides in Mexico and were/are able to make calls from CallManager SCCP phones to the Asterisk Server phones in the U.S.; however, we have not been able to call from Asterisk server in U.S. to CallManager phones in Mexico Here is what we tried: 1. Adding a Gatekeeper into CallManager and then have Asterisk (and also

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

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,

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

Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of asterisk@home installation. I have tried to use the oh323.conf content listed on WIKI but it is just not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot register. I need a working example of this file for similar phone. Seshu

ive had enough of cisco unity and microsoft exchange and im looking for alternatives to our voip system. right now, we have 3 cisco callmanagers, 1 cisco ip icd system, and 1 cisco unity voicemail system. all phones are cisco 7940/7960's and some ata186/188's. voice gateways are cisco vg200's with pri cards (5 total). im running h323 on the gateways and phones are of course

hello, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call:

Callmanager does nothing than construct and tear down calls and the actual RTP stream does not flow through the Callmanager but is direct from IP device to IP device. How does this work with Asterisk? I read something that lead me to believe that Asterisk has to process the entire call, is this the case? Blake Parker CCNA Network Engineer Alacare Home Health & Hospice, Inc. Email:

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

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

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

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some