Hi Alex,<br><br>You should create a dial plan to route sip calls to H.323 calls.<br><br>Take a look at :<br><a href="http://www.voip-info.org/wiki/">http://www.voip-info.org/wiki/</a><br><br><br><br><br><div><span class="gmail_quote">
On 8/6/07, <b class="gmail_sendername">Alessandro Russo</b> <<a href="mailto:ax.russo@gmail.com">ax.russo@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi to all,<br>I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper.<br>I've tested h323 using ohphone and I can talk between them, then I've tested SIP with Twinkle softphones and function very well.
<br>Now I have to perform call from h323 to sip and viceversa.<br>How can I do it ????<br>I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone: <br>- PSTN ==> CiscoVoiceGW(h323) ==> Asterisk ==> SIP
<br>- SIP ==> Asterisk ==> CiscoVoiceGW(h323) ==> PSNT<br><br>I've now idea how to configure asterisk (conf file) and softphones...<br>Thanks for all!<br clear="all"><span class="sg"><br>-- <br> AxR.
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