[Asterisk-Users] h323 how to ?????

[Ronald_Wiplinger]

I try to get H323 to run, but have so far only partial success:

There is a Gatekeeper GK, where asterisk connects to.

The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper.

 From the Network on the GK, asterisk is reachable via the number 
070333333. I have an extension on asterisk 6002, which is reachable.

I try to call a number attached to the gatekeeper (070168177) with the 
dialing plan:

exten => _9070.,1,Set(CALLERID(number)=070333333${CALLERIDNUM})
exten => _9070.,n,Dial(H323/${EXTEN:${TRUNKMSD}})
exten => _9070.,n,Hangup

CLI> shows:
*CLI>
    -- Executing Set("SIP/6002-9fac", "CALLERID(number)=0703333336002") 
in new stack
    -- Executing Dial("SIP/6002-9fac", "H323/070168177") in new stack
    -- Called 070168177
  == No one is available to answer at this time (1:0/0/0)
    -- Executing Hangup("SIP/6002-9fac", "") in new stack
  == Spawn extension (from-sip, 9070168177, 3) exited non-zero on 
'SIP/6002-9fac'

The gatekeeper sees nothing from that. I guess the syntax is wrong for 
dialing. How should it be?




Video connection:
I try to call from an H323 soft phone through the gatekeeper to call the 
extension 6003 (eyebeam)

H323 soft phone calls through GK Asterisk box without webcam installed:

    -- Executing Dial("H323/203.160.252.147-a44c", "SIP/8600") in new stack
Jul  8 13:51:37 WARNING[12674]: chan_sip.c:1742 create_addr: No such 
host: 8600
Jul  8 13:51:37 NOTICE[12674]: app_dial.c:977 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 3)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Answer("H323/203.160.252.147-a44c", "") in new stack
    -- Executing SetVar("H323/203.160.252.147-a44c", "TIMEOUT(digit)=5") 
in new stack
Jul  8 13:51:37 WARNING[12674]: pbx.c:5754 pbx_builtin_setvar_old: 
SetVar is deprecated, please use Set instead.
    -- Digit timeout set to 5
    -- Executing SetVar("H323/203.160.252.147-a44c", 
"TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing BackGround("H323/203.160.252.147-a44c", 
"demo-congrats") in new stack
    -- Playing 'demo-congrats' (language 'en')
  == CDR updated on H323/203.160.252.147-a44c
    -- Executing Dial("H323/203.160.252.147-a44c", "SIP/6003|60|trm") in 
new stack
    -- Called 6003
    -- Started music on hold, class 'default', on H323/203.160.252.147-a44c
    -- SIP/6003-e756 is ringing
    -- SIP/6003-e756 answered H323/203.160.252.147-a44c
    -- Stopped music on hold on H323/203.160.252.147-a44c
    -- Attempting native bridge of H323/203.160.252.147-a44c and 
SIP/6003-e756
Jul  8 13:52:16 WARNING[12674]: chan_sip.c:3203 process_sdp: Unknown SDP 
media type in offer: video 7156 RTP/AVP 105 34
Jul  8 13:52:16 WARNING[12674]: chan_h323.c:914 h323_indicate: Don't 
know how to indicate condition 17 on ooh323c_1
Jul  8 13:52:21 WARNING[12674]: chan_h323.c:914 h323_indicate: Don't 
know how to indicate condition 17 on ooh323c_1

No connection, not even audio!

sip.conf settings for 6003:

[6003]
type=friend
username=6003
secret=pwd
qualify=200
nat=yes
host=dynamic
canreinvite=yes
context=from-sip
callerid=Ronald Wiplinger <6003>	; Full caller
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=h261
allow=h263
allow=h263p 

Xten's settings:
Enable this SIP account
Display name:   Ronald at Leadtek
User name:      6003
Password: password
Authorization:  6003
Domain:         59.120.139.119

Domain Proxy:
x Register with domain

STUN server
x Manual override:    stun.xten.com


Any hints are welcome


bye

Ronald