<DIV>yeah yus<BR><BR><B><I>Yu Safin <calanet@gmail.com></I></B> wrote:
<BLOCKQUOTE class=replbq style="BORDER-LEFT: #1010ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: 5px">On 10/19/05, Steve Totaro <ASTERISK@TOTAROTECHNOLOGIES.COM>wrote:<BR>> YES<BR>><BR>> ----- Original Message -----<BR>> From: "Frank Kostin" <FRANKOSTIN@YAHOO.COM><BR>> To: <ASTERISK-USERS@LISTS.DIGIUM.COM><BR>> Sent: Wednesday, October 19, 2005 8:58 AM<BR>> Subject: [Asterisk-Users] SIP to IAX<BR>><BR>><BR>> Hello everybody,<BR>> Is it possible to route "any" incoming SIP call<BR>> (without authentication - register) from an Asterisk A<BR>> to a remote Asterisk B(throught IAX2), transparently ?<BR>> Otherwise said, I would like to pass any incoming SIP<BR>> call from Asterisk A to Asterisk B without SIP need to<BR>> be registered, like a phone call in zap.<BR>> I would apreciate any hint,<BR>> Thanks,<BR>> Frank<BR>><BR>short answer yes,<BR>read on,<BR>what you really need to know is the compression. You want to
avoid<BR>having to compress/uncompress different formats more than once. I<BR>normally have my SIP phones on 711 (same LAN to Asterisk A), then the<BR>calls travel via IAX2 to Asterisk B (yes, it is transparent). From<BR>Asterisk B, they might go to Zap phones, SIP phones, IAX phones, FXO<BR>(Zap), channel banks, etc.<BR>_______________________________________________<BR>--Bandwidth and Colocation sponsored by Easynews.com --<BR><BR>Asterisk-Users mailing list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE></DIV><p>
<hr size=1> <a href="http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=36035/*http://music.yahoo.com/unlimited/">Yahoo! Music Unlimited - Access over 1 million songs. Try it free.</a>