Asterisk 2011-11-01

athomehost*CLI> sip set debug on
SIP Debugging enabled
Reliably Transmitting (NAT) to 192.168.8.110:5060:
OPTIONS sip:101@192.168.8.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5bb9c4f0;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.8.1>;tag=as3e586d83
To: <sip:101@192.168.8.110:5060>
Contact: <sip:asterisk@192.168.8.1:5060>
Call-ID: 339f9d0c4a894d8a39607924731562a8@192.168.8.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4
Date: Tue, 01 Nov 2011 15:48:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


---

<--- SIP read from UDP:192.168.8.110:5060 --->
SIP/2.0 486 Busy Here
To: <sip:101@192.168.8.110:5060>;tag=a46b2f48cb3e8cb9i1
From: "asterisk" <sip:asterisk@192.168.8.1>;tag=as3e586d83
Call-ID: 339f9d0c4a894d8a39607924731562a8@192.168.8.1:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5bb9c4f0
Server: Linksys/SPA2102-5.1.5(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '339f9d0c4a894d8a39607924731562a8@192.168.8.1:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.8.110:5060 --->
INVITE sip:17772022880@192.168.8.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a
From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
To: <sip:17772022880@192.168.8.1>
Remote-Party-ID: L2SPA2102 <sip:101@192.168.8.1>;screen=yes;party=calling
Call-ID: d65495ce-e517150a@192.168.8.110
CSeq: 101 INVITE
Max-Forwards: 70
Contact: L2SPA2102 <sip:101@192.168.8.110:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.1.5(a)
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 5320193 5320193 IN IP4 192.168.8.110
s=-
c=IN IP4 192.168.8.110
t=0 0
m=audio 16430 RTP/AVP 18 0 2 4 8 96 97 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 20 lines) ---
Sending to 192.168.8.110:5060 (no NAT)
Using INVITE request as basis request - d65495ce-e517150a@192.168.8.110
Found peer '101' for '101' from 192.168.8.110:5060
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format G726-40 for ID 96
Found audio description format G726-24 for ID 97
Found audio description format G726-16 for ID 98
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.8.110:16430
Looking for 17772022880 in mario-default (domain 192.168.8.1)
list_route: hop: <sip:101@192.168.8.110:5060>

<--- Transmitting (NAT) to 192.168.8.110:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a;received=192.168.8.110;rport=5060
From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
To: <sip:17772022880@192.168.8.1>
Call-ID: d65495ce-e517150a@192.168.8.110
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:17772022880@192.168.8.1:5060>
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.8.110:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a;received=192.168.8.110;rport=5060
From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
To: <sip:17772022880@192.168.8.1>;tag=as0e6fcd3f
Call-ID: d65495ce-e517150a@192.168.8.110
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:17772022880@192.168.8.1:5060>
Content-Length: 0


<------------>

<--- Reliably Transmitting (NAT) to 192.168.8.110:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a;received=192.168.8.110;rport=5060
From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
To: <sip:17772022880@192.168.8.1>;tag=as0e6fcd3f
Call-ID: d65495ce-e517150a@192.168.8.110
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.8.110:5060 --->
ACK sip:17772022880@192.168.8.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.110:5060;branch=z9hG4bK-1c08ae3a
From: L2SPA2102 <sip:101@192.168.8.1>;tag=51d63efe2f05fd9ao1
To: <sip:17772022880@192.168.8.1>;tag=as0e6fcd3f
Call-ID: d65495ce-e517150a@192.168.8.110
CSeq: 101 ACK
Max-Forwards: 70
Contact: L2SPA2102 <sip:101@192.168.8.110:5060>
User-Agent: Linksys/SPA2102-5.1.5(a)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 'd65495ce-e517150a@192.168.8.110' Method: ACK
Really destroying SIP dialog '28a33b8611bed2be40140e0b4858ae2b@[c0a8:801:e82a:c97f::]' Method: REGISTER
athomehost*CLI>