asterisk users - Mar 2005 - how to sip->h323 using asterisk-oh323-0.7.1

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


	
		
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i am using gnugatekeeper. i have three things
gatekeeper ip, account, accountpassword how to set
account and password in oh323.conf

gatekeeper=gnu gatekeeper ip
gatekeeperPassword=accountpassword
accountCode=account

is this ok any example how to use this i want to rout
my sip call to this gatekeeper for h323.


	
		
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HELLO

i am using gungk gatekeeper from a provider. he has
given me a account,password,ip now i want to connect
to it with asterisk.

1. i want to call to my sip phones registered on my
local area network working. ok
2. i want to divert PSTN call to gun gatekeeper (from
service provider company). not working

the problem is that when i am trying to connect it
asterisk is desplaying message that Gatekeeper
'gatekeeper ip' found but faild to register.

i am using asterisk-oh323-0.7.1. one thing more when i
am using there diler to connection its working fine. 

oh323.conf
---------------
[general]
listenAddress=myip
listenPort=1720
connectPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=yes
h245Tunnelling=no
h254inSetup=no
inBandDTMF=yes
silenceSupperession=no
jitterMin=20
jitterMax=100
ipTos=none
tos=lowdelay
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
gatekeeper=provider ip
accountcode=account from provider
gatekeeperPassword=account password from provider
gatekeeperTTL=600
userUnputMode=TONE
amaFlags=default
context=default

[register]
context=default
alias=666

[665]
type=h323
prefix=321
context=default
codec=G711U
frames=20

extensions.conf
------------------
[default]
exten=>2000,1,Dial(SIP/${EXTEN})
exten=>3000,1,Dial(SIP/${EXTEN})
exten=>_321XXXXX,1,Dial(OH323:h323/${EXTEN@ipofprovider:1720|30|r)

2000, 3000 is working with 
i want 32145671 now my call should be transfered to
provider and dial his number 45671




		
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hello

now i am using my own gnugatekeeper. asterisk is
registering successfully with Gnugatekeeper. but it is
not transfering call to gnugk.

i am running 1234 user of OpenPhone with GNUgatekeeper
when i try to call from sip User agent 3000 to 3211234
asterisk is not forwarding it to GnuGK it replying
with 404 not found. 


gatekeeper.ini
----------------------------------------
[Gatekeeper::Main]
Fourtytwo=42
TimeToLive=600

[RoutedMode]
GKRouted=1
H245Routed=0
CallSignalPort=1721

[RasSrv::PermanentEndpoints]
asterisk mechine ip=xyz;123

[GkStatus::Auth]
rule=allow

on asterisk
oh323.conf
---------------
[general]
listenAddress=myip
listenPort=1719
connectPort=1719
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=yes
h245Tunnelling=no
h254inSetup=no
inBandDTMF=yes
silenceSupperession=no
jitterMin=20
jitterMax=100
ipTos=none
tos=lowdelay
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
gatekeeper=gnugk ip
accountcode=account
;gatekeeperPassword=account password
gatekeeperTTL=600
userUnputMode=TONE
amaFlags=default
context=default

[XYZ]
type=h323
prefix=123
context=default

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@gnugkip: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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I am just getting started using Asterisk and would like to know what ports I
need to open in my firewall for incoming and outgoing calls. I am running a
Cisco Pix 506 and I am having problems using Xlite to make calls through
Asterisk => Broadvoice and I think this maybe due to not having the proper
protocols passed since I can use X-lite on its own ok from home behind my
Linksys Router (no Xlite).

Thanks,
Scott

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@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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Hello,
Could anyone recommend something similar in functionality and
user-friendliness to SJPhone, but that would additionaly have IM and
presence support?

Thanks a lot,
Roman Zhovtulya