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
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