search for: h323phone

Displaying 4 results from an estimated 4 matches for "h323phone".

2005 May 13
3
2 minutes pause before ring on H323 channel
...e same for both SJPhone (soft phone) and QMix (PA168F). When I dial such extension I have to wait 2 minutes exactly (120 seconds) before extension rings. After long way of trial and errors with .conf files I managed to minimize this time to 1 minute exactly (60 seconds) exten => 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause before ring exten => 21,1,Dial(H323/h323phone@192.168.0.101) ; this leads to 60 seconds pause before ring After quick debugging session I found that this time goes to the call to H323EndPoint::MakeCallLocked(fullAddress, token, opts) in MyH323EndPoint::MakeC...
2004 May 08
0
H323 - Gatekeeper - asterisk - SIP config problems
...I lock asterisk to the other (eth0) IP - then I think this might work.... or must I put gnuGK on a separate machine? ps Documentation on the combination of Asterisk, h323 and Gatekeepers is really well hidden - I ain't seen it anywhere. In oh323.conf - I have the section... [register] context=h323phone alias=Call from gwprefix=0 gwprefix=1 gwprefix=2 gwprefix=3 gwprefix=4 gwprefix=5 gwprefix=6 gwprefix=7 1 - [h323phone] in extensions.conf is identical to my [sip] section (for my internal phones) - seems to work OK. 2 - the 'Call from' appears now with the CLID on the displays of the H323...
2004 Dec 07
1
H.323 trunking
Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the H.323phone (61-8004) gives me the following error -- Executing Dial("SIP/2004-8350", "H323/192.168.204.13...
2004 Aug 06
0
Urgent help with Sip <------> H323 on FREEBSD
I need some help with getting the following to work SipPhone <------> Asterisk <------> H323 GK (quintum) And H323Phone <------> Asterisk <------> H323 GK (quintum) I have tried to run the Asterisk from the newest ports and could after some digging around in the configs register the SipPone to Asterisk and Asterisk to the H323 GK. But when I try to make a call from The SipPhone I get Segmentation faul...