droplet_404

droplet_404

dur*CLI> 
dur*CLI> sip set debug on
SIP Debugging enabled
dur*CLI> 
dur*CLI> channel originate SIP/thufir extension 18003569377@outbound
  == Using SIP RTP CoS mark 5
Audio is at 19288
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to <my.external.ip.address>:5060:
INVITE sip:thufir@192.168.1.5:5062 SIP/2.0
Via: SIP/2.0/UDP <<droplet.floating.ip>>:5060;branch=z9hG4bK4fff4bef
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as285c7b45
To: <sip:thufir@192.168.1.5:5062>
Contact: <sip:anonymous@<<droplet.floating.ip>>:5060>
Call-ID: 3c33e3ac0d9ed3030eb02d397089367a@<<droplet.floating.ip>>:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Wed, 06 Jul 2016 09:50:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 300

v=0
o=root 1319064596 1319064596 IN IP4 <<droplet.floating.ip>>
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 <<droplet.floating.ip>>
t=0 0
m=audio 19288 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:<my.external.ip.address>:5060 --->
SIP/2.0 404 Not Found
To: <sip:thufir@192.168.1.5:5062>;tag=470632d69d55b230i0
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as285c7b45
Call-ID: 3c33e3ac0d9ed3030eb02d397089367a@<<droplet.floating.ip>>:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP <<droplet.floating.ip>>:5060;branch=z9hG4bK4fff4bef
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to <my.external.ip.address>:5060:
ACK sip:thufir@192.168.1.5:5062 SIP/2.0
Via: SIP/2.0/UDP <<droplet.floating.ip>>:5060;branch=z9hG4bK4fff4bef
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as285c7b45
To: <sip:thufir@192.168.1.5:5062>;tag=470632d69d55b230i0
Contact: <sip:anonymous@<<droplet.floating.ip>>:5060>
Call-ID: 3c33e3ac0d9ed3030eb02d397089367a@<<droplet.floating.ip>>:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog '3c33e3ac0d9ed3030eb02d397089367a@<<droplet.floating.ip>>:5060' in 6400 ms (Method: INVITE)
dur*CLI> 
dur*CLI> sip set debug off
SIP Debugging Disabled
dur*CLI> 
dur*CLI>