-- Executing [3012082228@npt-agent:1] Dial("SIP/2008-0000000e", "SIP/3012082228@itc-tpx-outbound") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14800
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 65.60.126.253:5060:
INVITE sip:3012082228@smartvoice.telepacific.com SIP/2.0
Via: SIP/2.0/UDP 192.168.15.108:5060;branch=z9hG4bK7757af4b
Max-Forwards: 70
From: "2008" <sip:9092937640@smartvoice.telepacific.com>;tag=as372750d3
To: <sip:3012082228@smartvoice.telepacific.com>
Contact: <sip:9092937640@192.168.15.108:5060>
Call-ID: 48867f947c218f1277467b1b01d7381d@smartvoice.telepacific.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.0.1
Date: Thu, 07 Feb 2019 19:33:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "2008" <sip:2008@smartvoice.telepacific.com>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 1216806648 1216806648 IN IP4 192.168.15.108
s=Asterisk PBX 16.0.1
c=IN IP4 192.168.15.108
t=0 0
m=audio 14800 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/3012082228@itc-tpx-outbound

<--- SIP read from UDP:65.60.126.253:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.108:5060;branch=z9hG4bK7757af4b
From: "2008" <sip:9092937640@smartvoice.telepacific.com>;tag=as372750d3
To: <sip:3012082228@smartvoice.telepacific.com>
Call-ID: 48867f947c218f1277467b1b01d7381d@smartvoice.telepacific.com
CSeq: 102 INVITE

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

<--- SIP read from UDP:65.60.126.253:5060 --->
SIP/2.0 604 Does Not Exist Anywhere
Via: SIP/2.0/UDP 192.168.15.108:5060;branch=z9hG4bK7757af4b
From: "2008" <sip:9092937640@smartvoice.telepacific.com>;tag=as372750d3
To: <sip:3012082228@smartvoice.telepacific.com>;tag=52635000-1549568033829
Call-ID: 48867f947c218f1277467b1b01d7381d@smartvoice.telepacific.com
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 604 "Does Not Exist Anywhere" back from 65.60.126.253:5060
Transmitting (no NAT) to 65.60.126.253:5060:
ACK sip:3012082228@smartvoice.telepacific.com SIP/2.0
Via: SIP/2.0/UDP 192.168.15.108:5060;branch=z9hG4bK7757af4b
Max-Forwards: 70
From: "2008" <sip:9092937640@smartvoice.telepacific.com>;tag=as372750d3
To: <sip:3012082228@smartvoice.telepacific.com>;tag=52635000-1549568033829
Contact: <sip:9092937640@192.168.15.108:5060>
Call-ID: 48867f947c218f1277467b1b01d7381d@smartvoice.telepacific.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.0.1
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [3012082228@npt-agent:2] Hangup("SIP/2008-0000000e", "") in new stack
  == Spawn extension (npt-agent, 3012082228, 2) exited non-zero on 'SIP/2008-0000000e'
Really destroying SIP dialog '48867f947c218f1277467b1b01d7381d@smartvoice.telepacific.com' Method: INVITE