Hi, thanks for reply I'm reading more about Dialplan, but until now, I've not found anything...(like example or tutorial) With the word "route" you are intending the "Goto" command?? Please spent some minutes for helping me ^_^ If you are agree, I send you some information about configuration files. Thx
On 8/6/07, map <[EMAIL PROTECTED]> wrote: > > 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 <[EMAIL PROTECTED]> 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 > > > > > _______________________________________________ > --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 > -- Alessandro R.
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