<--- SIP read from UDP:172.30.42.2:19360 --->
INVITE sip:7203@172.30.245.208 SIP/2.0
Via: SIP/2.0/UDP 172.30.42.2:19360;branch=z9hG4bK-d87543-7a109024624c8a15-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:7207@172.30.42.2:19360>
To: "7203"<sip:7203@172.30.245.208>
From: "Satish Patel"<sip:7207@172.30.245.208>;tag=ab276854
Call-ID: OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 283

v=0
o=- 5 2 IN IP4 172.30.42.2
s=CounterPath X-Lite 3.0
c=IN IP4 172.30.42.2
t=0 0
m=audio 18196 RTP/AVP 0 8 101
a=alt:1 2 : 0GaPOjkj 4s+pKS4i 172.30.42.2 18196
a=alt:2 1 : bkgQVdlj YPYWayP4 192.168.1.106 18196
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Sending to 172.30.42.2:19360 (no NAT)
Using INVITE request as basis request - OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.
Found peer '7207' for '7207' from 172.30.42.2:19360
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/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 172.30.42.2:18196
Looking for 7203 in from-sip (domain 172.30.245.208)
list_route: hop: <sip:7207@172.30.42.2:19360>

<--- Transmitting (no NAT) to 172.30.42.2:19360 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.42.2:19360;branch=z9hG4bK-d87543-7a109024624c8a15-1--d87543-;received=172.30.42.2;rport=19360
From: "Satish Patel"<sip:7207@172.30.245.208>;tag=ab276854
To: "7203"<sip:7203@172.30.245.208>
Call-ID: OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.3.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:7203@172.30.245.208:5060>
Content-Length: 0


<------------>
    -- Executing [7203@from-sip:1] Macro("SIP/7207-00000002", "stdexten,7203,sip/7203") in new stack
    -- Executing [s@macro-stdexten:1] Dial("SIP/7207-00000002", "sip/7203,10,t") in new stack
  == Using SIP RTP CoS mark 5
[Apr 28 19:31:25] ERROR[10604]: netsock2.c:263 ast_sockaddr_resolve: getnameinfo: Name or service not known
[Apr 28 19:31:25] WARNING[10604]: chan_sip.c:5182 create_addr: No such host: 7203
Really destroying SIP dialog '370df7ce27817a822e399d1972c40bdd@[::1]:0' Method: INVITE
[Apr 28 19:31:25] WARNING[10604]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-stdexten:2] Goto("SIP/7207-00000002", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-stdexten,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-stdexten:1] Goto("SIP/7207-00000002", "s-NOANSWER,1") in new stack
    -- Goto (macro-stdexten,s-NOANSWER,1)
    -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/7207-00000002", "7203,u") in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.30.42.2:19360 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.42.2:19360;branch=z9hG4bK-d87543-7a109024624c8a15-1--d87543-;received=172.30.42.2;rport=19360
From: "Satish Patel"<sip:7207@172.30.245.208>;tag=ab276854
To: "7203"<sip:7203@172.30.245.208>;tag=as67ee6916
Call-ID: OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.3.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:7203@172.30.245.208:5060>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 535374663 535374663 IN IP4 172.30.245.208
s=Asterisk PBX 1.8.3.3
c=IN IP4 172.30.245.208
t=0 0
m=audio 17196 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Apr 28 19:31:25] WARNING[10604]: app_voicemail.c:5535 leave_voicemail: No entry in voicemail config file for '7203'
    -- Executing [s-NOANSWER@macro-stdexten:2] Goto("SIP/7207-00000002", "default,s,1") in new stack
    -- Goto (default,s,1)
  == Channel 'SIP/7207-00000002' jumping out of macro 'stdexten'
    -- Sent into invalid extension 's' in context 'default' on SIP/7207-00000002
    -- Executing [i@default:1] Playback("SIP/7207-00000002", "invalid") in new stack
    -- <SIP/7207-00000002> Playing 'invalid.ulaw' (language 'en')

<--- SIP read from UDP:172.30.42.2:19360 --->
ACK sip:7203@172.30.245.208:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.42.2:19360;branch=z9hG4bK-d87543-5c0c75116e473478-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:7207@172.30.42.2:19360>
To: "7203"<sip:7203@172.30.245.208>;tag=as67ee6916
From: "Satish Patel"<sip:7207@172.30.245.208>;tag=ab276854
Call-ID: OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.
CSeq: 1 ACK
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Executing [i@default:2] Hangup("SIP/7207-00000002", "") in new stack
  == Spawn extension (default, i, 2) exited non-zero on 'SIP/7207-00000002'
Scheduling destruction of SIP dialog 'OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.' in 9472 ms (Method: ACK)
set_destination: Parsing <sip:7207@172.30.42.2:19360> for address/port to send to
set_destination: set destination to 172.30.42.2:19360
Reliably Transmitting (no NAT) to 172.30.42.2:19360:
BYE sip:7207@172.30.42.2:19360 SIP/2.0
Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK622ab2d9;rport
Max-Forwards: 70
From: "7203"<sip:7203@172.30.245.208>;tag=as67ee6916
To: "Satish Patel"<sip:7207@172.30.245.208>;tag=ab276854
Call-ID: OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.3.3
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 20
Content-Length: 0


---

<--- SIP read from UDP:172.30.42.2:19360 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.245.208:5060;branch=z9hG4bK622ab2d9;rport=5060
Contact: <sip:7207@172.30.42.2:19360>
To: "Satish Patel"<sip:7207@172.30.245.208>;tag=ab276854
From: "7203"<sip:7203@172.30.245.208>;tag=as67ee6916
Call-ID: OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.
CSeq: 102 BYE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'OTY2YjQxZGUwYmY0YzRmODcxNTM0Yzk3NDk3MDI5NzY.' Method: ACK

*CLI>
<--- SIP read from UDP:172.30.42.2:19360 --->


<------------->