[Asterisk-Users] Asterisk OH-323 module-Inbound Call dropped due to
in-call-rate violation (1.55)
Bukoka Budoka
budoka806 at hotmail.com
Wed Nov 9 11:52:49 MST 2005
Hi to all,
i have installed the latest CVS asterisk version as well as the
asterisk-oh323-0.7.3.
I have also installed the openh323-v1_17_2 and pwlib-v1_9_1 ( i also tried
the Mimas patched oh323 and pwlib but they did not behave well as far as the
gatekeeper registration was concerned).
The problem i now have, is that when i call from a h323 terminal
(netmeeting) to an Asterisk registered SIP client i get the following:
Nov 9 18:19:56 WARNING[20122] chan_oh323.c: Inbound call
'ip$192.168.1.1:10235/23826-488a9126' dropped due to in-call-rate violation
(1.55)
--->where 192.168.1.1 is the asterisk server.
The oh323.conf is as follows:
; 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
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; 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
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
;outboundMax=100
;inboundMax=100
;simultaneousMax=100
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=20000
;crlThreshold=30
;
; 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=10
libTraceLevel=10
libTraceFile=/var/log/asterisk/oh323.log
;
; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is the
zone name.
; Valid values for this option are:
; DISABLE,
; DISCOVER,
; <gatekeeper's DNS name>,
; <gatekeeper's ip>,
; GKID:<gatekeeper's id>
; <gatekeeper's id>@<gatekeeper's name or address>
;
gatekeeper=192.168.2.1
;gatekeeper=DISCOVER
;
; Set the gatekeeper password. If used, it enables H.235 access to
gatekeeper.
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout. Before the expiration of
; the timeout, a re-registration is attempted.
;
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
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Default language
;
language=en
;
; Default Music-On-Hold class
;
musiconhold=default
;
; Set the default context of H.323 calls.
;
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
gw=12345678
;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
;-----------------------------------------
; 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=G711A
frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
;codec=G729
;frames=2
Have you seen such a message before?
Budoka
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