Michael,
Yeah.. for sure the channel is loaded.. calling to my asterisks works fine.
I have included the oh323.conf and the original message.
Thanks a lot for you help. I would would like to get this baby working.
Alex
The log;
Nov 8 18:04:01 WARNING[294930]: channel.c:1901 ast_request: No channel type registered for 'OH323'
Nov 8 18:04:01 NOTICE[294930]: app_dial.c:742 dial_exec: Unable to create channel of type 'OH323'
Extensions.conf exten => 495234,3,Dial(OH323/192.168.1.20)
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=no ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no ; ; 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=none ; ; 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=TONE ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; 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]. ; ; ; 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 On 8-nov-04, at 11:09, Michael Manousos wrote:
Since you are able to receive H.323 calls with chan_oh323, I assume that the module is loaded. You could check the incoming/outgoing/simultaneous limits or submit the oh323.conf. Additionally, what are the full messages that you get on the console?
Michael.
Alex van Es wrote:Hi all,
For my setup I need to forward incoming SIP and ZAP calls to my IP phone using H323. I have managed to set up the OH323 and when I enter my asterisk's ip number into sjphone, it will answer and give me the welcome message. So receiving calls with H323 is not a problem.. but I want to be able to dial out.
I have set up a extention that looks like;
exten => 1234,1,Dial(OH323/192.168.1.20)
I keep on getting the message unable to create channel of type ' OH323'. I have tried also the names h323, h.323, oh323, OH323/h323.. but none of them seem to exist. When I receive the incoming call it says channel OH323, so I assume that is the correct name. However.. I still can't forward calls out.
I could do without OH323, but when I forward incoming SIP calls to my IP phone using SIP I just get silence after I answer the phone (both parties can't hear each other) so I wanted to try it this way.
Anyone has any ideas?
Alex
--
Alex van Es - [EMAIL PROTECTED]
http://photography.icepick.com
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-- Alex van Es - [EMAIL PROTECTED] http://photography.icepick.com
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