Help Forwarding incoming DID TO SIP URI

Need some guidence
I want to forward incoming DID to a sip uri
Eg , Sip/[email protected]
It’s on a brand new clean install 32 bit
I can see the notice in asterisk from the inbound calls but that’s as far I have got

Create an extension of type custom, and in the dial field use the URI noted above in the format:
SIP/[email protected]:5060
Then direct calls to this extension with an inbound route.

Thanks for the info I will try that and let you know how i get on

hi tried this and did not work the pbx plays a message number dialed has not been recognized

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] NoOp(“SIP/”, “Received incoming SIP connection from unknown peer to 435667887”) in new stack
– Executing [[email protected]:2] Set(“SIP/”, “DID=675646456”) in new stack
– Executing [[email protected]:3] Goto(“SIP/”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [[email protected]:1] GotoIf(“SIP/”, “0?checklang:noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [[email protected]:5] Set(“SIP/”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2015-06-12 09:20:54.788 CEST.
– Executing [[email protected]:6] Log(“SIP/”, "WARNING,“Rejecting unknown SIP connection from"”) in new stack
[2015-06-12 09:20:39] WARNING[4194][C-00000007]: Ext. s:6 @ from-sip-external: “Rejecting unknown SIP connection from”
– Executing [[email protected]:7] Answer(“SIP/”, “”) in new stack
– Executing [[email protected]:8] Wait(“SIP/”, “2”) in new stack
[2015-06-12 09:20:39] NOTICE[4194][C-00000007]: channel.c:4301 __ast_read: Dropping incompatible voice frame on SIP/ of format ulaw since our native format has changed to (alaw)
– Executing [[email protected]:9] Playback(“SIP/”, “ss-noservice”) in new stack
– <SIP/> Playing ‘ss-noservice.alaw’ (language ‘en’)
== Spawn extension (from-sip-external, s, 9) exited non-zero on ‘SIP/’
– Executing [[email protected]:1] Hangup(“SIP/”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/’

Surprised at the lack of answers on this subject as it has massive opertunities for pbx servers
It opens the doors and possibilities to any pbx system
I know it can be done because I know some one that’s done it , but he’s not available to help at this moment
I’ve been doing some research and it seems that it’s not just a case of making changes via the web interface
But also editing some of the files
This can only be done from the root login but I will be looking into FTP also
FTP would then let me down load the file make a copy and then upload it back

I will be working on this extensively and post results