Hi,
I have about one year of experience with Asterisk, working with ZAP (digium, junghanns) ZAPHFC, SIP and IAX. These technologies are quite clear to me, the problem is that I have no experience with H323, but now, I need to use this also.
The problem that I have is very trivial, so I think that this should be a very easy question for you guys whom know how it works.
All I want to do, is use a H323 phone, SJPhone on my Asterisk. I have compiled the H323 of asterisk, i.e. not OH323. With the configuration below, I can make a call from my H323 phone, make it enter in it's context in the dialplan (from-h323 in my h323.conf). So in this direction all is ok. My problem is the other direction, calling with my SIP phone, I'm not able to make the H323 phone ring. Instead Asterisk tells me "no one is available to answer at this time", but if I've called my SIP phone seconds before, it works (?!).
I'd be really happy if someone could give me a simple, working h323.conf, and the correct dial syntax for extensions.conf.
Tim
h323.conf
[general] port = 1720 bindaddr = 0.0.0.0 context=h323 disallow=all allow=alaw gatekeeper=DISABLE [114] type=user context=from-h323 host=192.168.1.164
extensions.conf exten => _2.,1,Dial(H323/[EMAIL PROTECTED])
asterisk says:
-- Executing Dial("SIP/116-94e6", "H323/[EMAIL PROTECTED]") in new stack
16:41:01.344 ThreadID=0x441d4bb0 h323ep.cxx(1323) H323 Making call to: [EMAIL PROTECTED]
-- Called [EMAIL PROTECTED]
16:41:11.345 H225 Caller:815b200 transports.cxx(1587) H323TCP Could not connect to 192.168.1.164:1720 (local port=0) - Connection timed out(110)
16:41:11.345 H225 Caller:815b200 h323.cxx(1445) H225 Sending release complete PDU: callRef=10466
16:41:11.347 H323 Cleaner h323.cxx(1542) H323 Connection ip$localhost/10466 terminated.
== No one is available to answer at this time
-- Executing Hangup("SIP/116-94e6", "") in new stack
== Spawn extension (from-sip, 22, 2) exited non-zero on 'SIP/116-94e6'
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