== Manager 'ami_guy' logged on from 127.0.0.1
    -- Called 501@originate/n
    -- Executing [501@originate:1] NoOp("Local/501@originate-00000000;2", "") in new stack
    -- Executing [501@originate:2] NoCDR("Local/501@originate-00000000;2", "") in new stack
    -- Executing [501@originate:3] Dial("Local/501@originate-00000000;2", "SIP/501") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 12120
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.22.200.164:5060:
INVITE sip:501@172.22.200.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK18cf35e9
Max-Forwards: 70
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
To: <sip:501@172.22.200.164:5060>
Contact: <sip:2482481234@172.22.1.100:5060>
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.0.0
Date: Mon, 17 Nov 2014 22:14:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1820360807 1820360807 IN IP4 172.22.1.100
s=Asterisk PBX 13.0.0
c=IN IP4 172.22.1.100
t=0 0
m=audio 12120 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/501

<--- SIP read from UDP:172.22.200.164:5060 --->
SIP/2.0 401 Unauthorized
To: <sip:501@172.22.200.164:5060>;tag=201676ed804ce1f5i0
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK18cf35e9
Server: Linksys/PAP2-3.1.22(LS)
WWW-Authenticate: Digest realm="172.22.1.100", nonce="38c0cdcd", qop="auth", algorithm=md5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 172.22.200.164:5060:
ACK sip:501@172.22.200.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK18cf35e9
Max-Forwards: 70
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
To: <sip:501@172.22.200.164:5060>;tag=201676ed804ce1f5i0
Contact: <sip:2482481234@172.22.1.100:5060>
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0


---
Audio is at 12120
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.22.200.164:5060:
INVITE sip:501@172.22.200.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK6cc7693a
Max-Forwards: 70
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
To: <sip:501@172.22.200.164:5060>
Contact: <sip:2482481234@172.22.1.100:5060>
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.0.0
Authorization: Digest username="501", realm="172.22.1.100", algorithm=MD5, uri="sip:501@172.22.200.164:5060", nonce="38c0cdcd", response="ccd1042f0bc1fd855331a0f56a824405", qop=auth, cnonce="3a25a94f", nc=00000001
Date: Mon, 17 Nov 2014 22:14:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1820360807 1820360808 IN IP4 172.22.1.100
s=Asterisk PBX 13.0.0
c=IN IP4 172.22.1.100
t=0 0
m=audio 12120 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

---

<--- SIP read from UDP:172.22.200.164:5060 --->
SIP/2.0 100 Trying
To: <sip:501@172.22.200.164:5060>
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 103 INVITE
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK6cc7693a
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 172.22.200.164:5060:
CANCEL sip:501@172.22.200.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK6cc7693a
Max-Forwards: 70
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
To: <sip:501@172.22.200.164:5060>
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060' in 6400 ms (Method: INVITE)
  == Spawn extension (originate, 501, 3) exited non-zero on 'Local/501@originate-00000000;2'

<--- SIP read from UDP:172.22.200.164:5060 --->
SIP/2.0 180 Ringing
To: <sip:501@172.22.200.164:5060>;tag=addfb2071bcf5e9i0
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 103 INVITE
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK6cc7693a
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
sip_route_dump: no route/path

<--- SIP read from UDP:172.22.200.164:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:501@172.22.200.164:5060>;tag=addfb2071bcf5e9i0
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 103 INVITE
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK6cc7693a
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 172.22.200.164:5060:
ACK sip:501@172.22.200.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK6cc7693a
Max-Forwards: 70
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
To: <sip:501@172.22.200.164:5060>;tag=addfb2071bcf5e9i0
Contact: <sip:2482481234@172.22.1.100:5060>
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.0.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:172.22.200.164:5060 --->
SIP/2.0 200 OK
To: <sip:501@172.22.200.164:5060>;tag=addfb2071bcf5e9i0
From: <sip:2482481234@172.22.1.100>;tag=as30a4bc2b
Call-ID: 647d430968f454c155d91f4e2f0d6fb0@172.22.1.100:5060
CSeq: 103 CANCEL
Via: SIP/2.0/UDP 172.22.1.100:5060;branch=z9hG4bK6cc7693a
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0

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