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Asterisk talking to Alcatel OmniPCX PBX using H323

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SYSTEM
Asterisk@Home 1.1
CentOS release 3.4 (final)
Linux version 2.4.21-27.0.1.EL
Wildcard FXO: Generic Clone (for meetme conference hardware timing!!! if fxo is not present then bad conference.)

Alcatel OmniPCX software version 4 (only H323). SIP protocol only from version 5…

APPLICATION
meetme conference incoming call (SIP phones, ZAP FXO, Alcatel OmniPCX H323)
User call OmniPCX, OmniPCX call asterisk via H323 at number 9999, asterisk receive call in from-internal context…
9999 is menu program for conference select.

Last Problem :
If conference is opened by user from Alcatel OmniPCX then only this user receive bad audio…if user recall no problem. Conference is good. Conference is not very stable. Please wait network testing…

H323 TALKING

Asterisk@Home
Self install file asteriskathome-h323-1.0.zip
– asterisk-oh323-0.6.5
– gnugk-2.2.1-linux-x86
– openh323
– pwlib


edit /etc/asterisk/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=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=yes
;
; 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=10.xxx.xxx.xxx      <-----IP of Alcatel OmniPCX

;
; 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-internal

;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
alias=CONFERENCE           <-----------NAME DISPLAYED on Alcatel telephone  
alias=3000                            <-----------NUMBER internal on Alcatel telephone

;
; 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=10
;codec=G711A
;frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
;codec=G729
;frames=2

Alcatel OmniPCX
Shop IP license for h323 communication (2 license 1000euro)

ARS Table

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Asterisk talking to Alcatel OmniPCX PBX using H323 - Voip-info.org
Asterisk talking to Alcatel OmniPCX PBX using H323 - Voip-info.org - 1

 

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