[Asterisk-Users] Forward incoming SIP calls to H323 ipphone?

Alex van Es Alex at icepick.com
Mon Nov 8 09:54:00 MST 2004


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 - Alex at icepick.com
>> http://photography.icepick.com
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
--
Alex van Es - Alex at icepick.com
http://photography.icepick.com




More information about the asterisk-users mailing list