Hi,
Maybe this is off topic, but I thought there is somebody who can help me
in this list.
I'm trying to configure asterisk-1.0.7 to work with gnugk-2.0.8.
Something like:
SIP phones -> ASTERISK -> GNUGK ->Cisco GW -> PSTN
|
h323 phones
Following is h323.conf in Asterisk:
--------------------------------------------------------------------------------
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=g729
gatekeeper = x.x.x.x
secret = 1234
AllowGKRouted = yes
noFastStart = yes
noH245Tunneling = yes
noSilenceSuppression = yes
[30598272]
type=h323
prefix=1100001,1100005,1100006,1100007
context=home
;e164=1100007
[1100005]
type=user
context=home
incominglimit=4
--------------------------------------------------------------------
sip.conf in asterisk
--------------------------------------------------------------------
[general]
port=5060 ; Port to
bind to
bindaddr=0.0.0.0
; Address to bind SIP channel to
context=home ; Default context
for incoming calls
musicclass=default
;videosupport=yes
allow=g729
allow=g723
;externip = 202.179.0.164
;localnet=
192.168.0.0/255.255.0.0
[1100001]
type=friend
username=1100001
;secret=1111
host=dynamic
nat=yes
defaultip=192.168.0.11
context=home
canreinvite=no
callerid=1100001
mailbox=1100001@local
[1100002]
type=friend
username=1100002
;secret=2222
nat=yes
host=dynamic
context=home
canreinvite=no
callerid=1100002
mailbox=1100002@local
[1100005]
type=friend
username=1100005
;secret=1234
defaultip=192.168.0.62
nat=yes
host=dynamic
context=home
canreinvite=no
callerid=1100005
mailbox=1100005@local
[1100006]
type=friend
username=1100006
;secret=4321
host=dynamic
context=home
canreinvite=no
callerid=1100006
mailbox=1100006@local
--------------------------------------------------------------------------------
As in above configuration I'm registering Asterisk as an endpoint to
gnugk.
It is working and I can make calls from SIP phones to PSTN.
However my question is, how can I call from h323 endpoints to SIP
phones or vice versa in above case?
Is it possible? I'm afraid, it can't since asterisk is itself an one
endpoint to gnugk.
If possible how can I make it work?
If not, is it possible to register or make each SIP phones to be known to
gnugk?
How can I accomplish that? Ideally this solution could be the
best.
It would be very helpful if somebody can show me the config samples, or
give me some hints and advices.
thanks in advance,
Ganbold
