similar to: Register Asterisk with Gatekeeper - oh323

Dear friends, I have compiled and installed h.323 in my asterisk. And I have a service from a H.323 VoIP provider who give me user, password and gatekeeper IP address. All configured. But when I start my asterisk i receive the following error and h.323 calls can not be making and/or receiving. [chan_h323.so]=> (The NuFone Network's Open H.323 Channel Driver) == Parsing

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong

Send these questions to Asterisk-Users mailing list. h323.conf ################################################## ; ; Configuration file of OpenH323 channel driver ; [general] listenAddress=W.X.Y.Z ; local ip listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=yes h245inSetup=yes jitterMin=20 jitterMax=100 ipTos=none outboundMax=100

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

Hi, I've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not

Hello there. Is there somebody with this connection working? I can't seem to make this work at all. Could someone please share some .conf files? Cheers, Vedran. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/71480bd9/attachment.htm

What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?

Hi, I am working on a H.323 project which involves GnuGk and Asterisk My current goal is to provide IVR functionality for the H.323 users which register through GnuGk(eg. call credit information) I have successfully built a H.323 platform using GnuGk - it uses SQL accounting and authorisation. Now I am trying to integrate it with Asterisk in order to provide IVR functionality as I already

Dear Mailinglist-User currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities. SIP and ISDN works fine, but H.323 not. In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeeper with the "chan_oh323" (version 0.6.5). We successfully tested in/egress calls without any problems. But when we started to connect our Asterisk

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

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, try to setup asterisk as an ISDN2H323-Gateway. The only problem i have after establishing a call is, that Audio works only from IP to ISDN-Phone but not from ISDN to IP-Phone. A known problem ??? Thanks in advance Michael i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus) and oh323-0.6.0 Here are my config's ############## # modem.conf # ##############

I'm trying to connect an asterisk server via oh323 to a Lucent iMerge. I patched the code due so that Lucent can handle asterisk's ver4 h323 http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration I can now successfully dial in to our company over multiple lines. The issue is when I dial out The first outgoing call connects to an outside user A The second call drops the first

hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &

Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

For the benefit of everyone, having H323 Configuration problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems