wlcpbx01*CLI>
wlcpbx01*CLI>
wlcpbx01*CLI>

<--- SIP read from UDP:192.168.100.51:5060 --->
INVITE sip:4848850228@192.168.4.121 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK0094eee6521fe111ad70e0ca94500775;rport
From: "tom soft phone" <sip:1231@192.168.4.121>;tag=1229236462
To: <sip:4848850228@192.168.4.121>
Call-ID: 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
CSeq: 27 INVITE
Contact: <sip:1231@192.168.100.51:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
ax-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:2159874491@192.168.4.121>
Content-Length:   417

v=0
o=- 782482750 0 IN IP4 192.168.100.51
s=SIPPER for PhonerLite
c=IN IP4 192.168.100.51
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:2263848821
a=sendrecv

<------------->
--- (14 headers 18 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 192.168.100.51 : 5060 (no NAT)
Using INVITE request as basis request - 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
Found peer '1231' for '1231' from 192.168.100.51:5060

<--- Reliably Transmitting (NAT) to 192.168.100.51:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK0094eee6521fe111ad70e0ca94500775;received=192.168.100.51;rport=5060
From: "tom soft phone" <sip:1231@192.168.4.121>;tag=1229236462
To: <sip:4848850228@192.168.4.121>;tag=as05097745
Call-ID: 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
CSeq: 27 INVITE
Server: FPBX-2.9.0(1.6.2.20)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33486f28"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.100.51:5060 --->
ACK sip:4848850228@192.168.4.121 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK0094eee6521fe111ad70e0ca94500775;rport
From: "tom soft phone" <sip:1231@192.168.4.121>;tag=1229236462
To: <sip:4848850228@192.168.4.121>;tag=as05097745
Call-ID: 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
CSeq: 27 ACK
Max-Forwards: 70
Content-Length: 0


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

<--- SIP read from UDP:192.168.100.51:5060 --->
INVITE sip:4848850228@192.168.4.121 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK0094eee6521fe111ad71e0ca94500775;rport
From: "tom soft phone" <sip:1231@192.168.4.121>;tag=1229236462
To: <sip:4848850228@192.168.4.121>
Call-ID: 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
CSeq: 28 INVITE
Contact: <sip:1231@192.168.100.51:5060>
Authorization: Digest username="1231", realm="asterisk", nonce="33486f28", uri="sip:4848850228@192.168.4.121", response="98f352650e54e9242fd57336e48ff103", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:2159874491@192.168.4.121>
Content-Length:   417

v=0
o=- 782482750 0 IN IP4 192.168.100.51
s=SIPPER for PhonerLite
c=IN IP4 192.168.100.51
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:2263848821
a=sendrecv

<------------->
--- (15 headers 18 lines) ---
Sending to 192.168.100.51 : 5060 (NAT)
Using INVITE request as basis request - 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
Found peer '1231' for '1231' from 192.168.100.51:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x1e0e (gsm|ulaw|alaw|g726|speex|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.100.51:5062
Looking for 4848850228 in from-internal (domain 192.168.4.121)
list_route: hop: <sip:1231@192.168.100.51:5060>

<--- Transmitting (NAT) to 192.168.100.51:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK0094eee6521fe111ad71e0ca94500775;received=192.168.100.51;rport=5060
From: "tom soft phone" <sip:1231@192.168.4.121>;tag=1229236462
To: <sip:4848850228@192.168.4.121>
Call-ID: 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
CSeq: 28 INVITE
Server: FPBX-2.9.0(1.6.2.20)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:4848850228@192.168.4.121>
Content-Length: 0


<------------>
    -- Executing [4848850228@from-internal:1] ResetCDR("SIP/1231-00000003", "") in new stack
    -- Executing [4848850228@from-internal:2] NoCDR("SIP/1231-00000003", "") in new stack
    -- Executing [4848850228@from-internal:3] Progress("SIP/1231-00000003", "") in new stack
Audio is at 192.168.4.121 port 10890
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 192.168.100.51:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK0094eee6521fe111ad71e0ca94500775;received=192.168.100.51;rport=5060
From: "tom soft phone" <sip:1231@192.168.4.121>;tag=1229236462
To: <sip:4848850228@192.168.4.121>;tag=as57b783fc
Call-ID: 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
CSeq: 28 INVITE
Server: FPBX-2.9.0(1.6.2.20)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:4848850228@192.168.4.121>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1055969520 1055969520 IN IP4 192.168.4.121
s=Asterisk PBX 1.6.2.20
c=IN IP4 192.168.4.121
t=0 0
m=audio 10890 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
    -- Executing [4848850228@from-internal:4] Wait("SIP/1231-00000003", "1") in new stack
    -- Executing [4848850228@from-internal:5] Progress("SIP/1231-00000003", "") in new stack
    -- Executing [4848850228@from-internal:6] Playback("SIP/1231-00000003", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <SIP/1231-00000003> Playing 'silence/1.ulaw' (language 'en')
    -- <SIP/1231-00000003> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
    -- <SIP/1231-00000003> Playing 'check-number-dial-again.gsm' (language 'en')
    -- Executing [4848850228@from-internal:7] Wait("SIP/1231-00000003", "1") in new stack
Reliably Transmitting (NAT) to 192.168.100.51:5060:
OPTIONS sip:1231@192.168.100.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.121:5060;branch=z9hG4bK4bf2c7a4;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.4.121>;tag=as50e60b42
To: <sip:1231@192.168.100.51:5060>
Contact: <sip:Unknown@192.168.4.121>
Call-ID: 0d0c8904321b2453069eb60811aefdd8@192.168.4.121
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.20)
Date: Wed, 07 Dec 2011 15:58:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.121:5060;branch=z9hG4bK4bf2c7a4;rport=5060
From: "Unknown" <sip:Unknown@192.168.4.121>;tag=as50e60b42
To: <sip:1231@192.168.100.51:5060>;tag=80de4bec521fe111ad71e0ca94500775
Call-ID: 0d0c8904321b2453069eb60811aefdd8@192.168.4.121
CSeq: 102 OPTIONS
Contact: <sip:1231@192.168.100.51:5060>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0d0c8904321b2453069eb60811aefdd8@192.168.4.121' Method: OPTIONS
    -- Executing [4848850228@from-internal:8] Congestion("SIP/1231-00000003", "20") in new stack

<--- Reliably Transmitting (NAT) to 192.168.100.51:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK0094eee6521fe111ad71e0ca94500775;received=192.168.100.51;rport=5060
From: "tom soft phone" <sip:1231@192.168.4.121>;tag=1229236462
To: <sip:4848850228@192.168.4.121>;tag=as57b783fc
Call-ID: 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
CSeq: 28 INVITE
Server: FPBX-2.9.0(1.6.2.20)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (from-internal, 4848850228, 8) exited non-zero on 'SIP/1231-00000003'
    -- Executing [h@from-internal:1] Hangup("SIP/1231-00000003", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1231-00000003'

<--- SIP read from UDP:192.168.100.51:5060 --->
ACK sip:4848850228@192.168.4.121 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK0094eee6521fe111ad71e0ca94500775;rport
From: "tom soft phone" <sip:1231@192.168.4.121>;tag=1229236462
To: <sip:4848850228@192.168.4.121>;tag=as57b783fc
Call-ID: 0094EEE6-521F-E111-AD6F-E0CA94500775@192.168.100.51
CSeq: 28 ACK
Authorization: Digest username="1231", realm="asterisk", nonce="33486f28", uri="sip:4848850228@192.168.4.121", response="98f352650e54e9242fd57336e48ff103", algorithm=MD5
Max-Forwards: 70
Content-Length: 0