On Thu, 2005-07-07 at 07:36 -0500, Guillermo Salas M
wrote:> 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.
>
I only heard the ringing tone on the H323 device, but no the voice or
something when the another party responds.
asterisk*CLI>
Configuration of OpenH323 channel driver
------------------------------------------
Version: 0.6.5
Listening on address: 0.0.0.0:1720
Gatekeeper used: ASTERISK@200.93.xxx.xx (Registered)
FastStart/H245Tunnelling/H245inSetup: ON/OFF/OFF
Supported formats in pref. order: g723<0> g729<1> alaw<2>
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 10000 - 10500
UDP (RAS) port range: 10000 - 20000
UDP (RTP) port range: 10000 - 20000
IP Type-of-Service value: 16
User input mode: 3
Max number of inbound H.323 calls: 10
Max number of outbound H.323 calls: 10
Max number of simultaneous H.323 calls: 10
Max call rate (ingress direction): 1.00/30
> 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 tel?fonos
> 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.
>
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--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP : 103@sip.manta.telconet.net
e-mail: gsalas@manta.telconet.net
www : http://www.telconet.net
http://www.telcocarrier.net
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