[Asterisk-Users] Asterisk + GNUGK

Ganbold Tsagaankhuu ganbold at gmail.com
Thu May 5 17:36:01 MST 2005


Hi,

On 5/5/05, Niksa Baldun <niksa.baldun at lumiss.hr> wrote:
> Assuming your h.323 phones are registered with gnugk, you need to
> instruct gnugk to forward certain numbers to Asterisk. In OH323 (which I
> am using) you would need to add something like:
> 
> [register]
> gwprefix=0
> gwprefix=1
> etc.
> 
> In h323.conf, I believe you have to add prefix=xxx in your endpoint
> definition.

I tried to put prefixes in h323.conf in following way, however it didn't work:

[30598272]
type=h323
prefix=1100001
prefix=1100005
prefix=1100006
prefix=1100007
context=home
;e164=1100007
--------------------------------------------------------------------------

Am I doing something wrong? Does somebody have configuration samples?

thanks,

Ganbold


> 
> Bear in mind though that H.323 support in Asterisk is rather inadequate
> (only basic telephony functions are available).
> 
> Niksa Baldun
> 
> 
> Ganbold Tsagaankhuu wrote:
> 
> >Hi,
> >
> >I'm trying to configure asterisk to work with gnugk-2.0.8. Something like:
> >
> >SIP phones -> ASTERISK -> GNUGK ->Cisco GW -> PSTN
> >                                              |
> >                                           h323 phones
> >
> >
> >Following is h323.conf:
> >--------------------------------------------------------------------------------
> >[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
> >
> >[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 at local
> >
> >[1100002]
> >type=friend
> >username=1100002
> >;secret=2222
> >nat=yes
> >host=dynamic
> >context=home
> >canreinvite=no
> >callerid=1100002
> >mailbox=1100002 at 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 at local
> >
> >[1100006]
> >type=friend
> >username=1100006
> >;secret=4321
> >host=dynamic
> >context=home
> >canreinvite=no
> >callerid=1100006
> >mailbox=1100006 at 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.
> >
> >thanks in advance,
> >
> >Ganbold
> >_______________________________________________
> >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
> >
> >
> >
> >
>



More information about the asterisk-users mailing list