[Asterisk-Users] IConnectHere: Outbound working, Inbound not working....

[Oliver Brandt]

Found out why I was getting so little mail ... Had procmailrc only
watching for asterisk at marko.net mails and not for asterisk-users at lists.digium.com mails. Guess I got some catchin up to do...
Sorry for bothering you with this!
Oliver

On Mon, Feb 17, 2003 at 08:15:55PM +0100, Oliver Brandt wrote:
> Hi,
> 
> > Now I'm trying to get inbound calling working.  It seems that
> > when a call comes into Asterisk via iconnecthere it always falls
> > into the "s" extention.  I have two phone numbers with
> > iconnecthere and want to route calls to each of the numbers
> > differently.  Can this be done?  If so, how?
> > 
> > Basically I want callers to 1813342XXXX to be routed to 2102 and
> > calls to 1202454XXXX to be routed to 2103
> 
> I have not worke with incomming calls through iconnect so I'm not shure
> if this is what you need, but how about setting different contexts for
> your two iconnect numbers in your sip.conf? Then each should drop into
> its own context in the extensions.conf and you can do what ever you want
> with the calls.
> Hope this heped!
> Good luck!
> Oliver
> 
> > 
> > Thanks in advance for any help!
> > 
> > Here are my config files:
> > 
> > ;
> > ; SIP Configuration for Asterisk
> > ;
> > [general]
> > port = 5060			; Port to bind to
> > bindaddr = 0.0.0.0		; Address to bind to
> > context = default		; Default for incoming calls
> > register=1813342XXXX:XXXXXX at sipauth.deltathree.com 
> > register=1202454XXXX:XXXXXX at sipauth.deltathree.com 
> > 
> > [iconnecthere]
> > type=friend
> > username=XXXXXX
> > secret=XXXXXX
> > host=sipauth.deltathree.com
> > context=default
> > 
> > [2102]
> > type=friend
> > secret=XXXXXX
> > host=dynamic
> > context=default
> > 
> > [2103]
> > type=friend
> > secret=XXXXXX
> > host=dynamic
> > context=default
> > 
> > ;
> > ; Extention Configuration for Asterisk
> > ;
> > [default]
> > 
> > exten => 2102,1,Dial,SIP/2102 at 2102
> > exten => 2102,2,Voicemail(2102)
> > 
> > exten => 2103,1,Dial,SIP/2103 at 2103
> > exten => 2103,2,Voicemail(2103)
> > 
> > exten => _91XXXXXXXXXX,1,Dial,SIP/${EXTEN-1}@iconnecthere
> > exten => _91XXXXXXXXXX,2,Congestion
> > 
> > exten => s,1,Wait,1			; Wait a second, just for fun
> > exten => s,2,Answer			; Answer the line
> > exten => s,3,DigitTimeout,5		; Set Digit Timeout to 5 seconds
> > exten => s,4,ResponseTimeout,10		; Set Response Timeout to 10 seconds
> > exten => s,5,Directory,default
> > 
> > exten => t,1,Goto(#,1)			; If they take too long, give up
> > exten => i,1,Playback(invalid)		; "That's not valid, try again"
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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