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Diversion header

djsullie
Posts: 186
 Member Since:
2008-02-16
Submitted by djsullie on Wed, 11/19/2008 - 9:00am.

any ideas how I can add a Diversion: field to my sip invites? see below for an example.. Or how would I modify the Contact field

INVITE sip:+1xxxxxx@209.34.85.51:5060 SIP/2.0
Via: SIP/2.0/UDP 4.55.5.35:5060;branch=z9hG4bK02Bb3d32c5ddbef8d43
From: ;tag=gK0247c368
To:
Call-ID: 1627521408_6690@4.55.5.35
CSeq: 11532 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact:
P-Asserted-Identity:
Diversion: ;privacy=full;screen=no; reason=unknown; counter=1
Supported: 100rel
Content-Length: 298
Content-Disposition: session; handling=required
Content-Type: application/sdp

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Spirited
Posts: 11
 Member Since:
2009-06-22
Any update on this?I am
Mon, 06/22/2009 - 12:46pm

having same issue on FreePBX and Verizon


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joshelson
Posts: 244
 Member Since:
2006-12-07
I seem to recall doing this
Thu, 06/25/2009 - 7:05am

I seem to recall doing this with a modification to extensions_custom.conf.

Add the following lines (modified to reflect your dialing scheme as necessary) directly below the [from-internal-custom] context:

exten => _1NXXNXXXXXX,1,SipAddHeader(Diversion: <tel:866xxxxxxx>)
exten => _NXXNXXXXXX,1,SipAddHeader(Diversion: <tel: 866xxxxxxx>)
exten => _91NXXNXXXXXX,1,SipAddHeader(Diversion: <tel:866xxxxxxx>)
exten => _9NXXNXXXXXX,1,SipAddHeader(Diversion: <tel:866xxxxxxx>)

Do a sip debug in the Asterisk console, and you should see the diversion header in the SIP INVITE message.

Hope that helps,

Josh

--

FluentStream Technologies - Integrate * Communicate


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Spirited
Posts: 11
 Member Since:
2009-06-22
Not yet
Sat, 06/27/2009 - 12:49pm

I added in what you had (but changed the tel:866xxxxxxx to tel:xxxxxxxxxx
but no dice, after restart aster service I still have no diversion header, is there another place that you have code or macros?


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