<html><head><style type="text/css"><!-- DIV {margin:0px} --></style></head><body><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><div>Have you tried this guy's suggestion? (I have not, yet)<br><br><span><a target="_blank" href="http://slacker.com/%7Enugget/projects/asterisk/page7">http://slacker.com/~nugget/projects/asterisk/page7</a></span><br><br>--JW<br>----- Original Message ----<br>From: Joao Pereira <joao.pereira@fccn.pt><br>To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br>Sent: Thursday, April 6, 2006 7:44:56 AM<br>Subject: Re: [Asterisk-Users] Routing SIP calls via URI<br><br><div>But is there a way of doing this without a prefix?<br><br>because people should dial without prefixes: "name@domain.pt" , not like:<br>"6name@domain.pt"<br><br>How can we make this without a prefix? something
like:<br><br>if( !uri=~"@mydomain.pt" ){<br> forward the all to the Internet<br>}<br><br>:)<br>Thanks<br>Joao Pereira<br><br><br>Shad Mortazavi wrote:<br><br>>Dear Group,<br>><br>>I was able to fix this problem;<br>><br>>The solution was to use a prefix to dial out. <br>><br>>The next challenge was to send the SIP Domain over IAX2!. I found that<br>>if I included @SIPDOMAIN it would break the IAX2 communications.<br>><br>>exten => _6.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN}@SIPDOMAIN),<br>>breakes because @SIPDOMAIN is treated as the target context. You also<br>>can not include @Context after the @SIPDOMAIN.<br>><br>>I created a new variable DS which was a concatenation of EXTEN and<br>>SIPDOMAIN separated by % and not @ and I was now able to pass this over<br>>IAX2;<br>><br>>DS = EXTEN%SIPDOMAIN.<br>><br>>exten => _6.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${DS}).<br>><br>>At the other end I used
the CUT command and substring facilities in<br>>Asterisk to split DS by the % eliminator; I re-formed a new variable<br>>which was <br>><br>>DS = EXTEN@SIPDOMAIN.<br>><br>>I can now pass calls from my internal Asterisk server to my external<br>>Asterisk server using IAX2 and then call any external VoIP number.<br>><br>>Warm Regards<br>><br>>Shad Mortazavi<br>>------------------------------<br>>Nexus Group Technical Manager<br>>n|m Nexus Management Inc<br>><br>>-----Original Message-----<br>>From: Shad Mortazavi <br>>Sent: Thursday, March 30, 2006 10:30 AM<br>>To: asterisk-users@lists.digium.com<br>>Subject: Re: [Asterisk-Users] Routing SIP calls via URI<br>><br>>Dear Group;<br>><br>>I can confirm that I have read through the three examples in<br>>www.voip-info.org. <br>><br>>These examples are excellent and address a couple of the questions. I<br>>have IAX2 working between several asterisk
servers on our VPN and<br>>between the DMZ and our LAN. <br>><br>>Also<br>><br>>exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN})<br>><br>>This answers part of the question;<br>><br>>However what I want to do is to send any outbound sip calls via our<br>>external SIP server.<br>><br>>i.e;<br>> VPN LAN IAX2 DMZ Internet<br>>Internal UA <-------> Internal (*) <------> External (*)<------><br>>ExternalUA<br>><br>>We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX<br>>for Voicemail, 2xxx for Meetme, etc. <br>><br>>Do I need to setup a prefix to dial the internet? And then route all<br>>calls to the External(*) based on this
prefix?<br>><br>>Thanks<br>><br>>Shad Mortazavi<br>>------------------------------<br>>Nexus Group Technical Manager<br>>n|m Nexus Management Inc<br>><br>><br>>_______________________________________________<br>>--Bandwidth and Colocation provided by Easynews.com --<br>><br>>Asterisk-Users mailing list<br>>To UNSUBSCRIBE or update options visit:<br>> <a target="_blank" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>> <br>><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a target="_blank" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></div></div></div></div></body></html>