[Freeswitch-users] Diversion header

Jonas Gauffin jonas.gauffin at gmail.com
Tue Mar 2 08:20:26 PST 2010


Hello,

I got a incoming call from my provider which goes to a javascript. The
javascript transfers the call to an external destination (goes back through
my provider).
A diversion headers is added by FreeSWITCH on the outbound call. How can I
remove it (or if not possible, how do I change it)?

javascript:
        session.setCallerData("caller_id_number", aNumber);
session.setVariable("gate_caller_id_number", aNumber);
        session.setVariable("gate_site_id", SITE_ID);
        session.execute("transfer", this.dtmfBuffer + " XML internal");



send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200:
   ------------------------------------------------------------------------
   INVITE sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>SIP/2.0
   Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS
   Max-Forwards: 67
   From: "0236661XXX" <sip:0243795362 at 212.247.XX.XX>;tag=BHvytKtv47r2a
   To: <sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>
>
   Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5
   CSeq: 127654862 INVITE
   Contact: <sip:gw+dalaconnecttele2 at 212.247.XX.XX
:5060;transport=udp;gw=dalaconnecttele2>
   User-Agent: Gateon
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 273
   Diversion: 0 <sip:0 at 212.151.Y.Y
;user=phone>;reason=unconditional;counter=1;privacy=off
   X-FS-Support: update_display
   Remote-Party-ID: "0236661XXX" <sip:0243795362 at 212.247.XX.XX
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX
   s=FreeSWITCH
   c=IN IP4 212.247.XX.XX
   t=0 0
   m=audio 32712 RTP/AVP 8 3 101 13
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
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