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: [EMAIL PROTECTED] (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 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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html _______________________________________________ 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
