SIP Debugging Enabled for IP: 192.168.0.53

Audio is at 11962
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.53:5060:
INVITE sip:spa3000fxs@192.168.0.53:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK50ad5c48
Max-Forwards: 70
From: "8499...01" <sip:8499...01@192.168.0.1>;tag=as59b77c7e
To: <sip:spa3000fxs@192.168.0.53:5060>
Contact: <sip:8499...01@192.168.0.1:5060>
Call-ID: 7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 11 Dec 2014 00:42:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 233

v=0
o=root 503957875 503957875 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.32.0
c=IN IP4 192.168.0.1
t=0 0
m=audio 11962 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.0.53:5060 --->
SIP/2.0 100 Trying
To: <sip:spa3000fxs@192.168.0.53:5060>
From: "8499...01" <sip:8499...01@192.168.0.1>;tag=as59b77c7e
Call-ID: 7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK50ad5c48
Server: Linksys/SPA3000-3.1.20(GW)
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.53:5060 --->
SIP/2.0 180 Ringing
To: <sip:spa3000fxs@192.168.0.53:5060>;tag=429a3609f4cc7f8ci0
From: "8499...01" <sip:8499...01@192.168.0.1>;tag=as59b77c7e
Call-ID: 7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK50ad5c48
Server: Linksys/SPA3000-3.1.20(GW)
Remote-Party-ID: spa3000fxs <sip:spa3000fxs@192.168.0.1>;screen=yes;party=called
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: no route
Scheduling destruction of SIP dialog '7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.53:5060:
CANCEL sip:spa3000fxs@192.168.0.53:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK50ad5c48
Max-Forwards: 70
From: "8499...01" <sip:8499...01@192.168.0.1>;tag=as59b77c7e
To: <sip:spa3000fxs@192.168.0.53:5060>
Call-ID: 7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


---
Scheduling destruction of SIP dialog '7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.53:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:spa3000fxs@192.168.0.53:5060>;tag=429a3609f4cc7f8ci0
From: "8499...01" <sip:8499...01@192.168.0.1>;tag=as59b77c7e
Call-ID: 7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK50ad5c48
Server: Linksys/SPA3000-3.1.20(GW)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.53:5060:
ACK sip:spa3000fxs@192.168.0.53:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK50ad5c48
Max-Forwards: 70
From: "8499...01" <sip:8499...01@192.168.0.1>;tag=as59b77c7e
To: <sip:spa3000fxs@192.168.0.53:5060>;tag=429a3609f4cc7f8ci0
Contact: <sip:8499...01@192.168.0.1:5060>
Call-ID: 7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Scheduling destruction of SIP dialog '7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.53:5060 --->
SIP/2.0 200 OK
To: <sip:spa3000fxs@192.168.0.53:5060>;tag=429a3609f4cc7f8ci0
From: "8499...01" <sip:8499...01@192.168.0.1>;tag=as59b77c7e
Call-ID: 7eeef9a75c09f4a57a12b7724b4b8b6b@192.168.0.1:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK50ad5c48
Server: Linksys/SPA3000-3.1.20(GW)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
router*CLI> sip set debug off
SIP Debugging Disabled