How does a 4G VoLTE UE know the destination SIP URI format to create the SIP INVITE

This trace is the output from an ASR500 for a VoLTE call,
For VoLTE the UE and IMS core network must support Public User Identities as defined in section 13.4 of 3GPP TS 23.003, which includes all of the following types of addresses:
•Alphanumeric SIP-URIs
sip:[email protected]
•MSISDN represented as a SIP URI:
sip:[email protected];user=phone
•MSISDN represented as a Tel URI:
tel:+447700900123
sip:[email protected]
In the SIP SDP you will see: sip:[email protected]
Mobile Originating UE:        sip:[email protected]
Mobile Terminating UE:        tel:+14047808898
Notice the two different formats.....
Below in the initial SIP INVITE you will see that the MO (Mobile Originating) sends the SIP URI in the proper format (1 of 3) to the MT (Mobile Terminating 4G handset).
My questions is: does the MO know the SIP URI format of the MT (User Endpoint / 4G smartphone) because it has some sort of Address Book, or is that the designated format for a SIP INVITE   (to: tel+###########) because he MO knows the MSIDSN (tel number) dialed .
I do not understand how the MO knows how to format the SIP URI format of the MT (Mobile Terminating) and would appreciate any insight into this.
PROTOCOL PAYLOAD FOLLOWS:
2600:100c:8221:6dc9:f77a:8b7:5e38:a5d5.60717 > 2001:4888:3:fe0f:c0:105:0:17.5060: . [tcp sum ok] 1:1357(1356) ack 1 win 214 <nop,nop,timestamp 64706 423317258> (len 1388, hlim 64)
PROTOCOL PAYLOAD ENDS.
PDU HEX DUMP FOLLOWS:
0x0000 30ff 0594 c20d 0073 6000 0000 056c 0640 0            ......s`....l.@
0x0010 2600 100c 8221 6dc9 f77a 08b7 5e38 a5d5 &          ....!m..z..^8..
0x0020 2001 4888 0003 fe0f 00c0 0105 0000 0017                ..H.............
0x0030 ed2d 13c4 e245 e405 fcb6 417e 8010 00d6                .-...E....A~....
0x0040 f720 0000 0101 080a 0000 fcc2 193b 4f0a                 .............;O.
0x0050 494e 5649 5445 2074 656c 3a2b 3134 3034               INVITE.tel:+1404
0x0060 3738 3038 3839 3820 5349 502f 322e 300d                7808898.SIP/2.0.
0x0070 0a4d 6178 2d46 6f72 7761 7264 733a 2037                .Max-Forwards:.7
0x0080 300d 0a52 6f75 7465 3a20 3c73 6970 3a5b                0..Route:.<sip:[
0x0090 3230 3031 3a34 3838 383a 333a 6665 3066                2001:4888:3:fe0f
0x00a0 3a63 303a 3130 353a 3a31 375d 3a35 3036                :c0:105::17]:506
0x00b0 303b 6c72 3e0d 0a56 6961 3a20 5349 502f 0               ;lr>..Via:.SIP/
0x00c0 322e 302f 5443 5020 5b32 3630 303a 3130 2.               0/TCP.[2600:10
0x00d0 3063 3a38 3232 313a 3664 6339 3a66 3737                0c:8221:6dc9:f77
0x00e0 613a 3862 373a 3565 3338 3a61 3564 355d                a:8b7:5e38:a5d5]
0x00f0 3a35 3036 303b 6272 616e 6368 3d7a 3968                     :5060;branch=z9h
0x0100 4734 624b 3030 3033 3335 3933 2d31 6361                G4bK00033593-1ca
0x0110 6630 3633 340d 0a43 5365 713a 2031 2049                f0634..CSeq:.1.I
0x0120 4e56 4954 450d 0a46 726f 6d3a 203c 7369                NVITE..From:.<si
0x0130 703a 2b31 3931 3236 3735 3738 3639 4076                p:+19126757869@v
0x0140 7a69 6d73 2e63 6f6d 3e3b 7461 673d 3534                  zims.com>;tag=54
0x0150 3436 375f 3030 3033 3339 6130 2d33 6665                467_000339a0-3fe
0x0160 3439 3434 380d 0a54 6f3a 203c 7465 6c3a                49448..To:.<tel:
0x0170 2b31 3430 3437 3830 3838 3938 3e0d 0a41                +14047808898>..A

Hi Tod,
The "session target registrar " point to the SIP-TRUNK to the PSTN, as detailed exaplaination:
session target (VoIP dial peer)
To designate a network-specific address to receive calls from a VoIP or VoIPv6 dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command.
A ideal situation would be to use session target ipv4: of the ITSP:
dial-peer voice 105 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 91%...........
session protocol sipv2
session target ipv4:11.11.11.11:6034 <<(1st SIP-TRUNK)
voice-class codec 2
dtmf-relay rtp-nte
no vad
dial-peer voice 106 voip
description **Outgoing 2ND Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 91%...........
session protocol sipv2
session target ipv4:22.22.22.22:6035 <<(2ND SIP-TRUNK)
voice-class codec 2
dtmf-relay rtp-nte
no vad
Rate the post accordingly.
Regards,
Kevin

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      <JOB>MANAGER</JOB>
      <MGR>7839</MGR>
      <HIREDATE>1981-05-01</HIREDATE>
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      <DEPTNO>30</DEPTNO>
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      <DEPTNO>10</DEPTNO>
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      <DEPTNO>30</DEPTNO>
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<li> I handle only four datatypes here (VARCHAR2, NUMBER, DATE, TIMESTAMP). Of course you can add more.
The list of Oracle Type Number is available here : http://docs.oracle.com/cd/E11882_01/server.112/e26088/sql_elements001.htm#i54330
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