Error when attempting to dial to Cisco IOS peer's FXO ports to paging gw

gistfile1.txt

#### SIP DEBUG OUTPUT
    -- Executing [#201@hjuhsd_pbxin:1] Verbose("SCCP/4002-00000FB6", "Paging thru SPHS multipath") in new stack
Paging thru SPHS multipath
    -- Executing [#201@hjuhsd_pbxin:2] Dial("SCCP/4002-00000FB6", "SIP/sphs-page-gw,") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 28168
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.20.62.16:5060:
INVITE sip:172.20.62.16 SIP/2.0
Via: SIP/2.0/UDP 172.20.62.36:5060;branch=z9hG4bK6c792a89
Max-Forwards: 70
From: "Scott Pickle" <sip:4002@172.20.62.36>;tag=as350af0cd
To: <sip:172.20.62.16>
Contact: <sip:4002@172.20.62.36:5060>
Call-ID: 1630a9420659dce97a3ca24f3e92f109@172.20.62.36:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.1.5
Date: Sun, 21 Jan 2018 08:53:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Scott Pickle" <sip:4002@172.20.62.36>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 1988508724 1988508724 IN IP4 172.20.62.36
s=Asterisk PBX 15.1.5
c=IN IP4 172.20.62.36
t=0 0
m=audio 28168 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/sphs-page-gw

<--- SIP read from UDP:172.20.62.16:5060 --->
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 172.20.62.36:5060;branch=z9hG4bK6c792a89
From: "Scott Pickle" <sip:4002@172.20.62.36>;tag=as350af0cd
To: <sip:172.20.62.16>
Date: Sun, 11 Jul 1993 18:33:15 GMT
Call-ID: 1630a9420659dce97a3ca24f3e92f109@172.20.62.36:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.20.62.16:5060 --->
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 172.20.62.36:5060;branch=z9hG4bK6c792a89
From: "Scott Pickle" <sip:4002@172.20.62.36>;tag=as350af0cd
To: <sip:172.20.62.16>
Call-ID: 1630a9420659dce97a3ca24f3e92f109@172.20.62.36:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 172.20.62.16:5060:
ACK sip:172.20.62.16 SIP/2.0
Via: SIP/2.0/UDP 172.20.62.36:5060;branch=z9hG4bK6c792a89
Max-Forwards: 70
From: "Scott Pickle" <sip:4002@172.20.62.36>;tag=as350af0cd
To: <sip:172.20.62.16>
Contact: <sip:4002@172.20.62.36:5060>
Call-ID: 1630a9420659dce97a3ca24f3e92f109@172.20.62.36:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.1.5
Content-Length: 0


---
Scheduling destruction of SIP dialog '1630a9420659dce97a3ca24f3e92f109@172.20.62.36:5060' in 6400 ms (Method: INVITE)
  == Spawn extension (hjuhsd_pbxin, #201, 2) exited non-zero on 'SCCP/4002-00000FB6'
net-voip1*CLI> 

#sip.conf entry
[sphs-page-gw]
type=friend
context=inbound
qualify=yes
nat=no
insecure=port,invite
host=172.20.62.16 ; interfaces Valcom MultiPath 
dtmfmode=rfc2833
disallow=all
allow=ulaw,alaw

#dialplan entry
[sphs_valcom]
exten => #201,1,Verbose(Paging thru SPHS multipath)
 same => n,Dial(SIP/sphs-page-gw,20)
 same => n,Hangup()