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. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
