[asterisk-users] help: H323 and SIP

[map]

Hi Alex,

you should have a "route" for each extensions you would like to reach in
your extension.conf file.

Dial Plan is the main concept to understand in Asterisk.
Feel free to send you conf and I'll take a look.



On 8/6/07, Alessandro Russo <ax.russo at gmail.com> wrote:
>
> 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 < mapunt at gmail.com> 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 <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.
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>
>
> --
>
> Alessandro R.
> _______________________________________________
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>
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