[Asterisk-Users] Re: how to sip->h323 using asterisk-oh323-0.7.1

Charles Wang lazy.charles at gmail.com
Wed Mar 23 08:55:48 MST 2005


I also have problems the same with you.
I can find my asterisk registered on my GK's status port(7000).
And I make a call from my XPro to SIP and SIP to Asterisk, then
Asterisk calls to a H323 phone via GNUGK.

I can find the CDR message on GK's status monitor. But I the GK only
first ACF(tail with false) and unconnected CDR on my GK.

Do you solve your problem? Can you share me your asterisk config such
as extensions.conf and oh323.conf, gatekeeper.ini for me to refer? 
Please.

Best Regards
Charles


On Thu, 10 Mar 2005 23:33:53 -0800 (PST), Kamran Ahmad <p_kami at yahoo.com> wrote:
> hello
> 
> i am using my own gnugatekeeper as a gatekeeper for my
> asterisk. asterisk is registering successfully with
> Gnugatekeeper. but it is not transfering call to
> gnugk.
> 
> any one guide me who to do this
> --------------------------------------------------
> SJPhone(sipSoftPhone using sip)->asterisk
> asterisk(conversion from sip -> h.323)
> asterisk(send h.323)->GnuGK
> GnuGk->SoftPhone(h.323 OpenPhone)
> -------------------------------------------------
> 
> on GnuGatekeeper side
> gatekeeper.ini
> ----------------------------------------
> [Gatekeeper::Main]
> Fourtytwo=42
> TimeToLive=600
> 
> [RoutedMode]
> GKRouted=1
> H245Routed=0
> CallSignalPort=1721
> 
> [RasSrv::PermanentEndpoints]
> 192.168.0.203=xyz;123
> 
> [GkStatus::Auth]
> rule=allow
> 
> on asterisk
> oh323.conf
> ---------------
> ;
> ; Configuration file of OpenH323 channel driver
> ;
> 
> ;-----------------------------------------
> ; General configuration options
> ; (ports, jitter, GK, ...)
> ;-----------------------------------------
> [general]
> listenAddress=192.168.0.203
> listenPort=1719
> connectPort=1719
> 
> tcpStart=10000
> tcpEnd=20000
> 
> udpStart=10000
> udpEnd=20000
> 
> fastStart=yes
> 
> h245Tunnelling=no
> 
> h245inSetup=no
> 
> inBandDTMF=yes
> 
> silenceSuppression=no
> 
> jitterMin=20
> jitterMax=100
> 
> ipTos=none
> tos=lowdelay
> outboundMax=10
> inboundMax=10
> simultaneousMax=10
> 
> wrapLibTraceLevel=1
> libTraceLevel=1
> libTraceFile=stdout
> 
> gatekeeper=192.168.0.153
> gatekeeperPassword=test1
> accountcode=test1
> gatekeeperTTL=600
> 
> userInputMode=TONE
> 
> amaFlags=default
> 
> context=default
> 
> [xyz]
> type=h323
> prefix=123
> context=default
> 
> alias=1234
> context=default
> ;-----------------------------------------
> ; Specify and configure CODEC related
> ; options
> ;-----------------------------------------
> [codecs]
> codec=G711U
> frames=20
> 
> extensions.conf
> ------------------
> [default]
> exten=>2000,1,Dial(SIP/${EXTEN})
> exten=>3000,1,Dial(SIP/${EXTEN})
> exten=>_123XXXX,1,Dial(SIP/${EXTEN})
> exten=>_321XXXX,1,Dial(OH323:h323/${EXTEN at 192.168.0.153:1719|30|r)
> 
> sip.conf
> ------------------
> [2000]
> host=dynamic
> type=friend
> dtmfmode=INFO
> canreinvite=no
> 
> [3000]
> host=dynamic
> type=friend
> dtmfmode=INFO
> canreinvite=no
> 
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-- 

Best Regards
Charles



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