Bukoka Budoka
2005-Nov-09 11:52 UTC
[Asterisk-Users] Asterisk OH-323 module-Inbound Call dropped due to in-call-rate violation (1.55)
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 _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/
Bukoka Budoka
2005-Nov-10 02:46 UTC
[Asterisk-Users] Asterisk OH-323 module-Inbound Call dropped due to in-call-rate violation (1.55)
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 _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/