CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 91

Messages-Waiting: yes
Message-Account: sip:asterisk@89.35.79.5
Voice-Message: 1/0 (0/0)

---
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK789d7e43;rport
Call-ID: 25d802ee25621a3067d9900c56452cfe@89.35.79.5
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk@89.35.79.5>;tag=as4fa3de48
To: <sip:23001@89.35.79.247:5060>;tag=RvpIAcc0lRR8Kvub
Contact: <sip:23001@89.35.79.247:5060>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '25d802ee25621a3067d9900c56452cfe@89.35.79.5' Method: NOTIFY
voip*CLI> h
<--- SIP read from 89.35.79.69:16088 --->



<------------->
--- (0 headers 1 lines) ---
voip*CLI> help sip debug
Usage: sip set debug
       Enables dumping of SIP packets for debugging purposes

       sip set debug ip <host[:PORT]>
       Enables dumping of SIP packets to and from host.

       sip set debug peer <peername>
       Enables dumping of SIP packets to and from host.
       Require peer to be registered.
voip*CLI> sip set debug ip 10.0.0.1
SIP Debugging Enabled for IP: 10.0.0.1
voip*CLI> sip set debug off
SIP Debugging Disabled
voip*CLI> sip set debug ip 10.0.0.1
SIP Debugging Enabled for IP: 10.0.0.1
voip*CLI>
    -- Executing [0748881658@numberplan-custom-1:1] Dial("SIP/23003-08238cd8", "SIP/10.0.0.1/0748881658||T") in new stack
Audio is at 10.0.0.2 port 16096
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.1:5060:
INVITE sip:0748881658@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK63152fd7;rport
From: "Znuff" <sip:23003@10.0.0.2>;tag=as4fd86996
To: <sip:0748881658@10.0.0.1>
Contact: <sip:23003@10.0.0.2>
Call-ID: 71c8306f4e032f7220c0f1bd7ef9e366@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 06 Mar 2007 03:43:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 24756 24756 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 16096 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 10.0.0.1/0748881658
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 100 Trying
From: "Znuff"<sip:23003@10.0.0.2>;tag=as4fd86996
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-16c9d-590652c-552
Call-ID: 71c8306f4e032f7220c0f1bd7ef9e366@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK63152fd7;rport
Contact: <sip:0748881658@10.0.0.1>
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 415 Unsupported Media Type
From: "Znuff"<sip:23003@10.0.0.2>;tag=as4fd86996
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-16c9d-590652c-552
Call-ID: 71c8306f4e032f7220c0f1bd7ef9e366@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK63152fd7;rport
Content-Length:0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 415 "Unsupported Media Type" back from 10.0.0.1
Transmitting (no NAT) to 10.0.0.1:5060:
ACK sip:0748881658@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK63152fd7;rport
From: "Znuff" <sip:23003@10.0.0.2>;tag=as4fd86996
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-16c9d-590652c-552
Contact: <sip:23003@10.0.0.2>
Call-ID: 71c8306f4e032f7220c0f1bd7ef9e366@10.0.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0748881658@numberplan-custom-1:2] Congestion("SIP/23003-08238cd8", "") in new stack
  == Spawn extension (numberplan-custom-1, 0748881658, 2) exited non-zero on 'SIP/23003-08238cd8'
Really destroying SIP dialog '71c8306f4e032f7220c0f1bd7ef9e366@10.0.0.2' Method: INVITE
voip*CLI>