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hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

...st installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=yes silenceSuppression=no jitterMin=20 jitterMax=500 ipTos=reliability outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeep...

...Network's Open H.323 Channel Driver) == H.323 listener started *** Error registering with gatekeeper "200.69.192.252". Sep 29 00:07:32 ERROR[-151090528]: chan_h323.c:1987 load_module: Gatekeeper registration failed. my h323.conf is: [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=20...

...ports that are < 1024, # because some malicious (non-root) user may run a fake tincd on ports # above 1024. # The default port is 655, the port that has been assigned to tinc # by the IANA. If you want tincd to listen on any other port than 655, # you can use ListenPort for the `server', and ConnectPort for the # `client'. # You may use the prefixes 0x or 0 to denote a hexadecimal or octal # number respectively. ConnectPort = 655 # Listen on which port #ListenPort = 655 # Accept incoming connections AllowConnect = no # My own VPN IP # You may use the /nn notation to indicate the number of...

....xx.xx,xx,Tt) exten => 100,2,Hangup exten => 100,102,Hangup exten => 102,1,Dial(SIP/519,20,Tt) exten => 102,2,Hangup exten => 102,102,Hangup [local-access] include => extensions ------------- h323.conf ----------- [general] listenAddress=xx.xx.xx.xx,xx listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=DIS...

...jphone then every time whatever number I dial the call goes to asterisk and some computerized information regarding asterisk is coming. I am putting my h323.conf and ooh323.conf h323.conf ; The NuFone Network's ; Open H.323 driver configuration ; listenAddress=10.142.17.68 listenPort=1720 connectPort=1720 ;TCP tcpStart=10000 tcpEnd=20000 ;UDP udpStart=10000 udpEnd=20000 [general] port = 1720 bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for this machine ;tos=lowdelay ; ; You may specify a global default AMA flag for iaxtel calls. It must be ; one of 'default...

...------------------------------- [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 ; ;...

...ded because i want to make Voicemailboxes. If i connect via gnomemeeting to asterisk (call://3001@10.0.2.229) it works fine. Is there any way that i can send the called Number from CCM. I think it is sent but in a different way. My oh323.conf is: [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=yes h245inSetup=yes inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 wrapLibTraceLevel=1 libTraceLevel=1 libTraceFile=stdout gatekeeper=DISABLE userInpu...

...XXXXXX,1,SetVar(OH323_OUTCODEC=ulaw) exten => _1NXXNXXXXXX,2,Dial(OH323/00${EXTEN},60,r) ; international calls via h323 gatekeeper exten => _011.,1,SetVar(OH323_OUTCODEC=ulaw) exten => _011.,2,Dial(OH323/00${EXTEN}) Here is the oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=yes h245inSetup=no inBandDTMF=yes silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=100 inboundMax=100 simultaneousMax=100 ;bandwidthLimit=1024 wrapLibTraceLevel=1 libTraceLevel=0 libTrac...

...st to schedule in the past?!?! => (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found Aug 2 14:08:15 NOTICE[18873]: chan_oh323.c:4855 reload_config: Ignoring unknown H.323 [general] keyword 'connectPort', line 25. Aug 2 14:08:15 NOTICE[18873]: chan_oh323.c:4855 reload_config: Ignoring unknown H.323 [general] keyword 'silenceSuppression', line 58. Aug 2 14:08:15 NOTICE[18873]: res_musiconhold.c:490 monmp3thread: Request to schedule in the past?!?! [1]WrapH323EndPoint::WrapH323EndPoint...

Hello there. Is there somebody with this connection working? I can't seem to make this work at all. Could someone please share some .conf files? Cheers, Vedran. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/71480bd9/attachment.htm

...riteprotect=no [8540340] exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Dial(OH323/192.168.70.227) [voip-h323] exten => s,1,Answer exten => s,2,Dial,Modem/ttyI1:${OH323_DSTE164} ############## # oh323.conf # ############## [general] listenAddress=192.168.70.1 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=1 libTraceFile=stdout gatekeeper...

...esponse received on the Asterisk console sys.stderr.write("Received %s\n"%res) # And the obligatory flush sys.stderr.flush() ++++++++++++++++++++++++++++++++++++++++++++++++++ oh323.conf ++++++++++++++++++++++++++++++++++++++++++++++++++ [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=yes silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 bandwidthLimit=1024 wrapLibTraceLevel=1 libTraceLevel=1 libTraceFile=stdout gatekeeper=D...

...o(6) exten => s,4,GotoIf($[foo${OUTCID} = foo]?6) ;check for CID override for trunk exten => s,5,SetCallerID(${OUTCID}) exten => s,6,Dial(OH323/${ARG1}) exten => s,7,Congestion exten => s,107,Macro(outisbusy) and oh323.conf is general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=no h245inSetup=no inBandDTMF=yes silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=2 libTraceLevel=2 libTraceFile=stdout ; ip address...

...,1) exten => 4001,1,Dial(OH323/4001) exten => 4002,1,Dial(OH323/4002) exten => 4008,1,Dial(OH323/408) exten => 4122,1,Dial(SIP/4122,60,tr) exten => 4123,1,Dial(SIP/4123,60,tr) exten => _[5]ZXX,1,Dial(Modem/g1:${EXTEN}) oh323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=1024 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=s...

What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?

...ng@wol.be> -------------- next part -------------- ; ; Configuration file of OpenH323 channel driver ; ;----------------------------------------- ; General configuration options ; (ports, jitter, GK, ...) ;----------------------------------------- [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 libTraceLevel=1 libTraceFile=stdout ;gatekeeper=192.168.1.2 ;gatekeeper=DISCOVER ;gatekeeperPassword=secret gatekeeper=DISABLE ; ; Set the mo...

...UP BROADCAST RUNNING NOARP MULTICAST MTU:1500 Metric:1 RX packets:24 errors:0 dropped:0 overruns:0 frame:0 TX packets:118 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 Interrupt:5 client toncd.conf ---- ConnectTo = server.dns ConnectPort=8089 MyOwnVPNIP = 192.168.100.2/24 #VpnMask = 255.255.255.0 Passphrases=/usr/local/etc/tinc/passphrases/ TapDevice=/dev/tap0 client tap dev ---- tap0 Link encap:Ethernet HWaddr FE:FD:C0:A8:6F:02 inet addr:192.168.100.2 Bcast:192.168.100.255 Mask:255.255.255.0 UP BROADC...

...------------------------------- [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=30000 ;tcpEnd=65000 ; ; 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=30000 ;udpEnd=65000 ; ; Enable fast start (y...

hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &