<DIV>The reason i am using 1 scenario is because of routing, authentication and accounting. If i can use these things in asterisk i will use it.</DIV>
<DIV> </DIV>
<DIV>1)What is the best way to do that, through extensions ?</DIV>
<DIV> </DIV>
<DIV>2)When the call coming i can check the phone number and if the number should go to pstn i will forward this to asterisk. now i need example how i can forward the call to pstn in extension.cfg (or there are other way to forward the call to pstn inside asterisk).</DIV>
<DIV> </DIV>
<DIV>3) Another question if all my users authenticated with ser how i can send call to the other user who included in the same database and connected to the same SER server (LOCAL CALLS) should not go to the pstn. </DIV>
<DIV>Should i use softphone(user1) ->SER->ASTERISK->SER->softphone(user2)</DIV>
<DIV> or i need to use softphone(user1)->SER-(user2) if i use this scenario i will loose the cdrs of the asterisk.</DIV>
<DIV> </DIV>
<DIV>The reason i am asking the these question because till now i didn't use asterisk and i forwarded the call through ser and it's working fine. I wanted to use IVR system so i installed the asterisk and also asterisk has the CDRs. now i need to use this scenario </DIV>
<DIV>long distance call: softphone -> SER -> Asterisk -> pstn (long distance calls)</DIV>
<DIV>local calls: softphone->SER ->Asterisk -> SER->softphone ( I am not sure if i can do that without registering users inside sip.cfg in the asterisk.)</DIV>
<DIV> </DIV>
<DIV>Any help will be appreciated.</DIV>
<DIV> <BR><BR><B><I>Yair Hakak <yhakak@gmail.com></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Duh, i'm an idiot. I meant scenario #1.<BR><BR>-yair<BR><BR><BR>On Wed, 23 Mar 2005 18:52:28 +0200, Yair Hakak <YHAKAK@GMAIL.COM>wrote:<BR>> Hello,<BR>> what is the benefit of your scenario #2? I'm not understanding what<BR>> it adds for you...<BR>> <BR>> -yair<BR>> <BR>> <BR>> On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex <ALEXANDER_GAV@YAHOO.COM>wrote:<BR>> > Hi all<BR>> ><BR>> > I have a couple of questions maybe you guys can help me with them<BR>> ><BR>> > I have sip phones , SER server , Asterisk.<BR>> ><BR>> > what is the best way to do that (also with accounting and authentication).<BR>> ><BR>> > which one of those options<BR>> > 1) sipphone -> SER -> ASTERISK -> SER -> PSTN<BR>> ><BR>> > 2) sipphone -> SER ->ASTERISK ->PSTN<BR>> ><BR>> > on the
first option i am trying to return the call to the ser after it's<BR>> > pass the asterisk for some routing solutions and accounting. but i have some<BR>> > problems to hear the other side.<BR>> ><BR>> ><BR>> > Thanks for any advice<BR>> ><BR>> > ________________________________<BR>> > Do you Yahoo!?<BR>> > Yahoo! Small Business - Try our new resources site!<BR>> ><BR>> ><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>> ><BR>> ><BR>><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><p>
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