Ah that is brilliant, thanks a lot. Charles
On Mon, Jun 1, 2009 at 9:35 AM, Administrator TOOTAI <[email protected]>wrote: > Hi > > Charles Solar a écrit : > > Hi guys, I am new here but I have a quick question. > > > > I have an incoming trunk that sends me calls from various usernames I > have > > with them. Only trouble is they send invites as [email protected], not as > the > > username I have with them. So I cannot match extensions like I would > want > > to. > > Here is a sample invite > > > > INVITE sip:[email protected] SIP/2.0 > > Record-Route: <sip:0.0.0.0;lr=on;ftag=as29ffee59> > > Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK7238.25c90fc7.0 > > Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK1c709971;rport=5060 > > From: "" <sip:[email protected]>;tag=as29ffee59 > > To: <sip:[email protected]<sip%[email protected]>< > sip%[email protected] <sip%[email protected]> > >> > > Contact: <sip:[email protected]> > > Call-ID: 6a379af207d78b3b5f2e8c6c55e64009 > > CSeq: 102 INVITE > > User-Agent: Asterisk PBX > > Max-Forwards: 69 > > Date: Fri, 29 May 2009 04:12:09 GMT > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > Supported: replaces > > Content-Type: application/sdp > > Content-Length: 377 > > > > the only distinction between a call to username1 and username2 is in the > To: > > field, but I cannot find something to route the call based on the To > caller > > id. > > > > I think the dialednumber variable would be close to what I want, but > > apparently that is broken so I am unsure what to do. > > > [macro-setDialednumberFromSipHeader] > ; > ; We extract the DIALEDNUMBER from SIP header > ; which is of the form <sip:callednum...@ourasteriskipaddress> > > exten => s,1,Set(__DIALEDNUMBER=${SIP_HEADER(TO):5}) > exten => s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,@,1)}) > exten => s,n,GotoIf($["${DIALEDNUMBER:0:1}" != "+"]?numberIsOK) > exten => s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,+,2)}) > > exten => s,n(numberIsOK),NoOp() > exten => s,n,Set(CDR(dest)=${DIALEDNUMBER}) > > done ;-) > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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