[ASTERISK-140] res_pjsip_session: Add support for SIP-ISUP

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Add support for SIP-I (SIP-ISUP). Currently, it is not handled:

• Will likely need significant extensions within res_pjsip_session and possibly a new module as well
• Start with multipart handling... https://github.com/asterisk/asterisk/blob/master/res/res_pjsip_session.c#L3945

Normal, working call:

[Apr 25 22:54:13] Content-Disposition: session; handling=required
[Apr 25 22:54:13] Content-Type: application/sdp

Call that fails with 415:

[Apr 25 22:54:12] <--- Received SIP request (1464 bytes) from UDP:REDACTED:5060 --->
[Apr 25 22:54:12] INVITE sip:+1NPANXXXXXX@REDACTED:5060 SIP/2.0
[Apr 25 22:54:12] Via: SIP/2.0/UDP REDACTED:5060;branch=z9hG4bK0aB21100f1e9eae3b3e
[Apr 25 22:54:12] From: <sip:+CALLERNUMBER@REDACTED:5060;isup-oli=00>;tag=gK0a7123c5
[Apr 25 22:54:12] To: <sip:1NPANXXXXXX@REDACTED:5060>
[Apr 25 22:54:12] Call-ID: 206193188_67076456@REDACTED
[Apr 25 22:54:12] CSeq: 908783 INVITE
[Apr 25 22:54:12] Max-Forwards: 64
[Apr 25 22:54:12] Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
[Apr 25 22:54:12] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Apr 25 22:54:12] Contact: <sip:+CALLERNUMBER@REDACTED:5060>
[Apr 25 22:54:12] P-Asserted-Identity: <sip:+CALLERNUMBER;verstat=TN-Validation-Passed@REDACTED:5060>
[Apr 25 22:54:12] P-Charge-Info: sip:7717727505@REDACTED:5060
[Apr 25 22:54:12] Supported: timer,100rel,precondition
[Apr 25 22:54:12] Session-Expires: 1800
[Apr 25 22:54:12] Min-SE: 90
[Apr 25 22:54:12] Content-Length:   579
[Apr 25 22:54:12] Content-Type: multipart/mixed;boundary=sonus-content-delim
[Apr 25 22:54:12] MIME-Version: 1.0
[Apr 25 22:54:12]
[Apr 25 22:54:12] --sonus-content-delim
[Apr 25 22:54:12] Content-Disposition: session; handling=required
[Apr 25 22:54:12] Content-Length:   237
[Apr 25 22:54:12] Content-Type: application/sdp
[Apr 25 22:54:12]
[Apr 25 22:54:12] v=0
[Apr 25 22:54:12] o=Sonus_UAC 851117 540685 IN IP4 REDACTED
[Apr 25 22:54:12] s=SIP Media Capabilities
[Apr 25 22:54:12] c=IN IP4 REDACTED
[Apr 25 22:54:12] t=0 0
[Apr 25 22:54:12] m=audio 50064 RTP/AVP 0 101
[Apr 25 22:54:12] a=rtpmap:0 PCMU/8000
[Apr 25 22:54:12] a=rtpmap:101 telephone-event/8000
[Apr 25 22:54:12] a=fmtp:101 0-15
[Apr 25 22:54:12] a=sendrecv
[Apr 25 22:54:12] a=ptime:20
[Apr 25 22:54:12]
[Apr 25 22:54:12] --sonus-content-delim
[Apr 25 22:54:12] Content-Disposition: signal; handling=required
[Apr 25 22:54:12] Content-Type: application/isup; version=ansi88; base=ansi88
[Apr 25 22:54:12]
[Apr 25 22:54:12] `
...
[Apr 25 22:54:12] DEBUG[2757]: res_pjsip_session.c:4597 handle_outgoing_response:  TRUNKNAME: Method is INVITE, Response is 415 Unsupported Media Type
[Apr 25 22:54:12] DEBUG[2757]: res_pjsip_session/pjsip_session_reason_header.c:60 reason_header_outgoing_response:  TRUNKNAME: Response Code: 415
[Apr 25 22:54:12] DEBUG[2757]: res_pjsip_session/pjsip_session_reason_header.c:68 reason_header_outgoing_response:  TRUNKNAME: RC 415 not eligible for Reason header
[Apr 25 22:54:12] DEBUG[2757]: res_pjsip_session.c:4616 handle_outgoing_response:  TRUNKNAME
[Apr 25 22:54:12] <--- Transmitting SIP response (402 bytes) to UDP:REDACTED:5060 --->
[Apr 25 22:54:12] SIP/2.0 415 Unsupported Media Type
[Apr 25 22:54:12] Via: SIP/2.0/UDP REDACTED:5060;rport=5060;received=REDACTED;branch=z9hG4bK0aB21100f1e9eae3b3e
[Apr 25 22:54:12] Call-ID: 206193188_67076456@REDACTED
[Apr 25 22:54:12] From: <sip:+CALLERNUMBER@REDACTED;isup-oli=00>;tag=gK0a7123c5
[Apr 25 22:54:12] To: <sip:1NPANXXXXXX@REDACTED>;tag=122e6ffb-2dac-442b-ada8-3f837d645cf9
[Apr 25 22:54:12] CSeq: 908783 INVITE
[Apr 25 22:54:12] Server: Asterisk PBX 22.3.0
[Apr 25 22:54:12] Content-Length:  0

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