How to link Asterisks to a H323 gateway?

Comment on Asterisk oh323 channels
inimd
Joined: Fri 23 of Sep, 2005

How to link Asterisks to a H323 gateway?

Posted:Tue 23 of Jan, 2007 (17:33 UTC)
I just installed ooh323 0.73 in my asterisk box. oh323 commands can be found in asterisk CLI. But no calls can get through. Any one's help?

First problem, oh323.conf in etc/asterisk/ cannot be loaded with phpconfig. I need to restart amportal all the time. Does it mean I missed something during installation? What should I do?

Then my situation, what I need to do is:

SIP phones--->asterisk------>H323 soft switch for termination

The provider told me I can treat the soft switch as a gateway, gave me the IP, prefix and nothing else.

Then my oh323.conf

;
; Configuration file of OpenH323 channel driver
;

;-----------------------------------------
; General configuration options
; (ports, jitter, GK, ...)
;-----------------------------------------
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=10000
tcpEnd=20000
;
; Configure the UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
;       "rtp.conf"
;
udpStart=10000
udpEnd=20000
;
; Enable fast start (yes,no).
;
;fastStart=yes
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
;h245Tunnelling=yes
h245Tunnelling=no

;
; Enable early H.245 messages in call SETUP message.
;
;h245inSetup=yes
h425inSetup=no
;
; Set jitter buffer (in milliseconds, 20...10000).
;
jitterMin=20
jitterMax=100
;

ipTos=none


outboundMax=100
inboundMax=100
simultaneousMax=100


; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=3
libTraceLevel=3
libTraceFile=stdout
;

gatekeeper=DISABLE


gatekeeperTTL=600
;
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
;   Q931        -   Q.931 Keypad Information Element
;   STRING      -   H.245 string
;   TONE        -   H.245 tone
;   RFC2833     -   RFC2833
;   INBAND      -   
;
userInputMode=TONE

amaFlags=default

accountCode=H323

language=en

musiconhold=default

context=voip-h323

;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02

[codecs]
codec=G711A
frames=20

Then I add a line to extensions.conf [from-internal] for testing

exten => _9999X.,1,Dial(OH323/${EXTEN}@xxx.xxx.xxx.xxx:1720)

Then the debug:

    -- Executing Dial("SIP/9091-094537a8", "OH323/99998xxxxxx@xxx.xxx.xxx.xxx:1720") in new stack
    -- H.323 call to 99998xxxxxx@xxx.xxx.xxx.xxx:1720 with codec(s) alaw
    -- Outbound H.323 call to destination 99998xxxxxx@xxx.xxx.xxx.xxx:1720', channel 'OH323/99998xxxxxx@xxx.xxx.xxx.xxx:1720-3ee0f73'.
    -- Called 99998xxxxxx@xxx.xxx.xxx.xxx:1720

Then nothing happened. Even with a reason 0, 5,8,...until I hungup

    -- Hungup 'OH323/99998xxxxxx@xxx.xxx.xxx.xxx:1720'
  == Spawn extension (from-internal, 99998xxxxxx, 1) exited non-zero on 'SIP/9091-094537a8'