[asterisk-dev] Using Diversion: for follow-me w/ taqua switches?

[Philip A. Prindeville]

Has anyone had any luck using Diversion: with Taqua switches?

I've been trying to test it out by hand...  incoming calls to my
extension also ring on my cell phone, but unfortunately only give the
caller-id of our company's trunk #, not the original inbound call.

Jared helped me cobble up the following:

TRUNK=SIP/sip_proxy-out
...
CELL=Local/208xxxxxxxx at follow-me
PHILIP=SIP/bedroom_1&SIP/office_2&SIP/remote_2&SIP/kitchen_1&${CELL}
...

[redfish-extens]
exten => 111,1,Noop(Exten: ${EXTEN})
exten => 111,n,Macro(stdexten,111,${PHILIP},redfish)
exten => 111,n,Goto(s,exten)
...

; we assume that the extension is a 10-digit NANP number, and that we reach it via ${TRUNK}/${EXTEN}
; this is a fairly simply minded assumption, but it works for now in our trivial environment.

[follow-me]
exten => _NXXXXXXXXX,1,Log(NOTICE,Follow-me: ${EXTEN})
exten => _NXXXXXXXXX,n,Set(PEER=${CUT(GLOBAL(TRUNK)|/|2)})
exten => _NXXXXXXXXX,n,Set(IP=${SIPPEER(${PEER})})
exten => _NXXXXXXXXX,n,Set(URI=<sip:${CALLERID(num)}@${IP}>)
exten => _NXXXXXXXXX,n,SIPAddHeader(Diversion: ${QUOTE(${CALLERID(name)})} ${URI}\;reason=follow-me\;privacy=off)
; let the desk phones ring a bit before trying external lines...
exten => _NXXXXXXXXX,n,Wait(10)
exten => _NXXXXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN})



Alas, the switch barfs:

INVITE sip:208mmmmmmmm at y.y.y.y SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0d631355;rport
From: "Bob Barker" <sip:208nnnnnnn at x.x.x.x>;tag=as370ea433
To: <sip:208mmmmmmmm at y.y.y.y>
Contact: <sip:208nnnnnnn at x.x.x.x>
Call-ID: 368667cf4094cb4f53d30de5509dbad5 at x.x.x.x
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 Aug 2009 05:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Diversion: "Bob Barker" <sip:206oooooo at y.y.y.y>;reason=follow-me;privacy=off
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 26495 26495 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 17718 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

...
pbx*CLI> 
<--- SIP read from y.y.y.y:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0d631355;rport
From: "Bob Barker" <sip:208nnnnnnnn at x.x.x.x>;tag=as370ea433
To: <sip:208mmmmmmmm at y.y.y.y>
Call-ID: 368667cf4094cb4f53d30de5509dbad5 at x.x.x.x
CSeq: 102 INVITE
Content-Length: 0


Doh!!!!

Anyone else had any luck with this?  Is there anything bogus looking in my Diversion header?