I suggest you learn what spoofing does and how it works. There is a difference in spoofing and hijacking. When you spoof, you're pretending to be someone you're not.
You --> pretend to be 1.1.1.1 --> send traffic (easy to do)
Recipient --> responds to 1.1.1.1 --> this response will
NEVER get to you
But anyway, to make you understand why this won't work, I will now hurt your eyes with an explanation. If you can't understand based on the information I post here, I suggest you go read the RFCs on networking and SIP. Further, there is no absolute mechanism via the PSTN for someone to track an IP from a call. A carrier can, enduser can't. But here goes. So I decided to give you a breakdown of how the call would work and why it would fail.
The follow illustrates a call between extension_1000 (71.111.222.33) and extension_2000 (214.21.111.23) - the two addresses you used for your examples. You want to trick server C into thinking you are server B (214.21.111.23) So let's make this call:
Server B sends a SIP invite
214.21.111.23:5060 ->
71.111.222.33:5060
INVITE sip:extension1000@
71.111.222.33 SIP/2.0
CSeq: 1 INVITE
Via: SIP/2.0/UDP
214.21.111.23:5060
From: <sip:extension_2000@
214.21.111.23:5060>
Call-ID: spoofed_caller_id@
214.21.111.23To: <sip:extension1000@
71.111.222.33>
Contact: <sip:214.21.111.23:5060;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER
Content-Type: application/sdp
Content-Length: 228
Max-Forwards: 70
v=0
o=- xxxxx yyyyy IN
IP4 214.21.111.23s=SIL's Example
c=IN IP4
214.21.111.23t=0 0
m=audio 6268 RTP/AVP 18 101
a=
rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=
rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
Server C responds...
U
71.111.222.33:5060 ->
214.21.111.23:5060
SIP/2.0 200 OK
v: SIP/2.0/UDP
214.21.111.23:5060
f: <sip:extension_2000@
214.21.111.23:5060>
t: <sip:extension_1000@
71.111.222.33>
i: spoof_whatever_you_want_it_wont_work@
214.21.111.23CSeq: 1
INVITEm: <sip:extension_1000@
71.111.222.33:5060;transport=udp>
c: application/sdp
l: 177
v=0
o=- vvvvv xxxxx IN IP4
HOW_DO_YOU_PROPOSE_TO_INTERCEPT_AUDIO_FROM_ANOTHER_ADDRESSs=-
c=IN IP4
WHERE_IS_YOUR_MEDIA_PROXY_IN_THIS_MIX
t=0 0
m=audio 15960 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Server C responds: "Alright, let me make that extension ring" ...
U
71.111.222.33:5060 ->
214.21.111.23:5060
SIP/2.0 100 TryingCSeq: 1
INVITEv: SIP/2.0/UDP
214.21.111.23:5060
f: <sip:extension_2000@
214.21.111.23:5060>
i: spoofed_caller_id@
214.21.111.23t: <sip:extension_1000@
71.111.222.33>
l: 0
Server B acknowledges the call... "I'm ready!!!"
U
214.21.111.23:5060 ->
71.111.222.33:5060
ACK sip:extension_1000@
71.111.222.33:5060;transport=udp SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP
214.21.111.23:5060
From: <sip:extension_2000@
214.21.111.23:5060>
Call-ID: spoofed_caller_id@
214.21.111.23To: <sip:extension_2000@
71.111.222.33>
Contact: <sip:
214.21.111.23:5060;transport=udp>
Content-Length: 0
Max-Forwards: 70
Server C sets up audio RTP
U
71.111.222.33:5060 ->
214.21.111.23:5060
SIP/2.0 183 Session Progressv: SIP/2.0/UDP
214.21.111.23:5060
f: <sip:extension_2000@
214.21.111.23:5060>
t: <sip:extension_1000@
71.111.222.33>
i: spoofed_caller_id@
214.21.111.23
CSeq: 1 INVITE
m: <sip:extension_1000@
71.111.222.33:5060;transport=udp>
c: application/sdp
l: 177
v=0.
o=- 41201 4120100 IN IP4
HOW_WOULD_YOU_LIKE_TO_ADDRESS_RTP_ISSUESs=-
c=IN IP4
WHERE_IS_YOUR_MEDIA_PROXY_IN_THIS_MIXt=0 0
m=
audio 14612 RTP/AVP 18 101.
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
In the midst of this, you need to insert yourself between this connection to see this going on. Not on the same network? Good luck, you now have to hijack *something* to get inside that stream. May I suggest you go read some RFCs now. Understanding SIP and VoIP help more than spoofing. Even if you COULD hijack a session, what will you do for NAT, SRTP, TLS, and if the PBX has any redirects or proxy-auths?
And that concludes my post for the day

Sorry, work is overwhelming...