[asterisk-users] help: H323 and SIP

[map]

Hi Alex,

You should create a dial plan to route sip calls to H.323 calls.

Take a look at :
http://www.voip-info.org/wiki/




On 8/6/07, Alessandro Russo <ax.russo at gmail.com> wrote:
>
> Hi to all,
> I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper.
> I've tested h323 using ohphone and I can talk between them, then I've
> tested SIP with Twinkle softphones and function very well.
> Now I have to perform call from h323 to sip and viceversa.
> How can I do it ????
> I receive h323 call from a Cisco Voice GW to my Asterisk and this call
> have to go to a SIP phone:
> - PSTN ==> CiscoVoiceGW(h323) ==> Asterisk ==> SIP
> - SIP ==> Asterisk ==> CiscoVoiceGW(h323) ==> PSNT
>
> I've now idea how to configure asterisk (conf file) and softphones...
> Thanks for all!
>
> --
> AxR.
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070806/789749bc/attachment.htm