[Asterisk-Users] Calls with oh323 with no sound

[Guillermo Salas M]

Hi,

I've oh323 chan installed and working to make calls from SIP to H323
devices. The problem is can no hear sound with the H323 device. I think
this is some related with codecs o nat, because the H323 have one public
IP from a different subnet from the asterisk box.

If I use netmeeting in gateway mode, the call can be completed and I can
talk with a SIP device, but in gateway mode I can not call netmeeting
from SIP device.

This is the 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
;
; 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=20000
;
; 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=20000
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=no
;
; 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=no
;
; Set jitter buffer (in milliseconds, 20...10000).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
;
ipTos=lowdelay
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=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=0
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=DISCOVER
;
; 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=voip-h323
;context=from-pstn
context=from-internal

;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
; Colocar las extensiones SIP en esta seccion
alias=asterisk
; Para el Voice Mail
alias=*98
; Los teléfonos
alias=100
alias=101
alias=102
alias=103
alias=104
alias=105
alias=106
alias=107
alias=108
alias=109
alias=110
alias=200
alias=201
alias=202
alias=203
alias=204
alias=205
alias=206
alias=207
alias=208
alias=209
alias=210
alias=500
alias=501
alias=502
;
; 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

;-----------------------------------------
; 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)
;   G726        -   G.726(32k)
;   G72616K     -   G.726(16k)
;   G72624K     -   G.726(24k)
;   G72632K     -   G.726(32k)
;   G72640K     -   G.726(40k)
;   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=G711U
frames=20
codec=GSM0610
frames=4
codec=G7231
frames=2
codec=G729
frames=2
codec=G711A
frames=20

language=es

; EOF

Thank you.


Guillermo.