[Asterisk-Users] problems with using open h323

[C C]

Hi guys,
 
I have been trying for days to get openh323 connections to work on my asterisk box again.  About a week ago we started having problems with one-way audio on oh323 to zaptel calls after I did some software updating.  The calling party has no problems hearing the called party talking, but the called party cannot hear the calling party.
 
The incoming h323 calls are using the G729 codec with we bought from digium and I installed on this machine.
 
When I look at the asterisk CLI this is what I see:

voiper*CLI>
    -- Executing Dial("OH323/R2248", "Zap/g4/7005130218#") in new stack
    -- Called g4/7005130218#
    -- Zap/73-1 answered OH323/R2248
Mar 17 12:50:42 WARNING[507929]: chan_oh323.c:1400 oh323_read: OH323/R2248: Invalid format of RTP addresses.
    -- Hungup 'Zap/73-1'
  == Spawn extension (itl, 34917885138, 1) exited non-zero on 'OH323/R2248'
    -- Hungup 'OH323/R2248'
voiper*CLI>
 
 
Does anyone know why it might be saying "Invalid format of RTP addresses"?
 
Here is my oh323.conf file:
;
; 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
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Configure TCP port range to be used by H.323
;
tcpStart=10000
tcpEnd=50000
;
; Configure UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
;       "rtp.conf"
;
udpStart=10000
udpEnd=50000
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Enable silence suppression.
;
silenceSuppression=yes
;
; Set jitter buffer (in milliseconds, 20...10000).
;
jitterMin=20
jitterMax=1000
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;       lowdelay, throughput, reliability, mincost, none
;
ipTos=none
;
; Set the maximum number of inbound/outbound H.323 connections.
;
outboundMax=10
inboundMax=10
;
; 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 trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
;
; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
;       DISABLE,
;       DISCOVER,
;       <gatekeeper's DNS name>,
;       <gatekeeper's ip>,
;       GKID:<gatekeeper's id>
;
;gatekeeper=192.168.1.2
gatekeeper=DISABLE
;
; Set the gatekeeper password
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout
;
;gatekeeperTTL=600
;
; Set the mode for sending user-input
; Valid values for this option are:
;       Q931            -       Q.931 Keypad Information Element
;       STRING          -       H.245 string
;       TONE            -       H.245 tone
;       RFC2833         -       RFC2833
;
userInputMode=RFC2833
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Set the default context of H.323 calls.
;
 
context=itl
 
;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
[register]
;-----------------------------------------
; Specify and configure CODEC related
; options
;-----------------------------------------
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
;       G711U           -       G.711 u-Law
;       G711A           -       G.711 A-Law
;       G7231           -       G.723.1(6.3k)
;       G72316K3     -        G.723.1(6.3k)
;       G72315K3        -       G.723.1(5.3k)
;       G7231A6K3       -       G.723.1A(6.3k)
;       G7231A6K3       -       G.723.1A(6.3k)
;       G728            -       G.728
;       G729            -       G.729
;       G729A           -       G.729A
;       G729B           -       G.729B
;       G729AB          -       G.729AB
;       GSM0610         -       GSM 0610
;       MSGSM           -       Microsoft GSM Audio Capability
;       LPC10           -       LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;
;codec=G711A
;frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
codec=G729
codec=G729A
codec=G729B
codec=G729AB

Is there any information I left out?  Thanks if anyone has any ideas or can point me to some documentation that would be EXCELLENT!
 
 
Colin Clark

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