<--- SIP read from UDP:11.22.33.44:5060 --->
INVITE sip:+4412345@11.22.33.44;transport=UDP SIP/2.0
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes;did=733.29e7dc54>
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.0
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-cf7d037085b5fe20-1---d8754z-
Max-Forwards: 69
Contact: <sip:testphone@44.55.66.77:41890;transport=UDP>
To: <sip:+4412345@11.22.33.44;transport=UDP>
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
Allow-Events: presence, kpml
Content-Length: 300

v=0
o=Zoiper_user 0 0 IN IP4 192.168.5.1
s=Zoiper_session
c=IN IP4 44.55.66.77
t=0 0
m=audio 8000 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=direction:active
a=oldmediaip:192.168.5.1
<------------->
--- (28 headers 14 lines) ---
Sending to 11.22.33.44:5060 (no NAT)
Using INVITE request as basis request - YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
Found peer 'in.from.a' for 'in.from.a' from 11.22.33.44:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 44.55.66.77:8000
Looking for +4412345 in in.from.in.from.a (domain 11.22.33.44)
list_route: hop: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes;did=733.29e7dc54>

<--- Transmitting (NAT) to 11.22.33.44:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.0;received=11.22.33.44;rport=5060
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-cf7d037085b5fe20-1---d8754z-
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes;did=733.29e7dc54>
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
To: <sip:+4412345@11.22.33.44;transport=UDP>
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 2 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+4412345@88.99.11.22:5060>
Content-Length: 0


<------------>
    -- Executing [+4412345@in.from.in.from.a:1] NoOp("SIP/in.from.a-00000086", "in.from.in.from.a") in new stack
    -- Executing [+4412345@in.from.in.from.a:2] Dial("SIP/in.from.a-00000086", "SIP/+4412345@out.to.b,90,ieL(10800000)") in new stack
    -- Setting call duration limit to 10800.000 seconds.
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 17550
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 33.55.77.99:5060:
INVITE sip:+4412345@33.55.77.99 SIP/2.0
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK7964879d
Max-Forwards: 70
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>
Contact: <sip:+4466666@88.99.11.22:5060>
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 102 INVITE
User-Agent: Asteriskserver
Date: Mon, 24 Jun 2013 21:43:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
P-Asserted-Identity: <sip:+4466666@33.55.77.99:5060>
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 1707261853 1707261853 IN IP4 88.99.11.22
s=Asteriskserver SDP
c=IN IP4 88.99.11.22
t=0 0
m=audio 17550 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 SIP/+4412345@out.to.b

<--- SIP read from UDP:33.55.77.99:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK7964879d
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 102 INVITE
Server: Asteriskserver
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:33.55.77.99:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK7964879d
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE
Contact: <sip:33.55.77.99:5060>
Server: Asteriskserver
Content-Length: 208

v=0
o=- 3581099020 3581099020 IN IP4 22.44.66.88
s=-
c=IN IP4 22.44.66.88
t=0 0
m=audio 22122 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 10 lines) ---
list_route: hop: <sip:33.55.77.99:5060>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 22.44.66.88:22122
    -- SIP/out.to.b-00000087 is making progress passing it to SIP/in.from.a-00000086
Audio is at 21172
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 11.22.33.44:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.0;received=11.22.33.44;rport=5060
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-cf7d037085b5fe20-1---d8754z-
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes;did=733.29e7dc54>
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 2 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+4412345@88.99.11.22:5060>
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 1894142497 1894142497 IN IP4 88.99.11.22
s=Asteriskserver SDP
c=IN IP4 88.99.11.22
t=0 0
m=audio 21172 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

<------------>

<--- SIP read from UDP:33.55.77.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK7964879d
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Supported: replaces, 100rel, timer
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE
Session-Expires: 1800;refresher=uas
Require: timer
Contact: <sip:33.55.77.99:5060>
Server: Asteriskserver
Content-Length: 208

v=0
o=- 3581099020 3581099020 IN IP4 22.44.66.88
s=-
c=IN IP4 22.44.66.88
t=0 0
m=audio 22122 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 10 lines) ---
list_route: hop: <sip:33.55.77.99:5060>
set_destination: Parsing <sip:33.55.77.99:5060> for address/port to send to
set_destination: set destination to 33.55.77.99:5060
Transmitting (no NAT) to 33.55.77.99:5060:
ACK sip:33.55.77.99:5060 SIP/2.0
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK2a7104f2
Max-Forwards: 70
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Contact: <sip:+4466666@88.99.11.22:5060>
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 102 ACK
User-Agent: Asteriskserver
Content-Length: 0


---
    -- SIP/out.to.b-00000087 answered SIP/in.from.a-00000086
Audio is at 21172
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 11.22.33.44:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.0;received=11.22.33.44;rport=5060
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-cf7d037085b5fe20-1---d8754z-
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes;did=733.29e7dc54>
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 2 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+4412345@88.99.11.22:5060>
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 1894142497 1894142498 IN IP4 88.99.11.22
s=Asteriskserver SDP
c=IN IP4 88.99.11.22
t=0 0
m=audio 21172 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

<------------>
    -- Locally bridging SIP/in.from.a-00000086 and SIP/out.to.b-00000087

<--- SIP read from UDP:11.22.33.44:5060 --->
ACK sip:+4412345@88.99.11.22:5060 SIP/2.0
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes>
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKb637.1eca5954.2
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-872e94aaac46ca38-1---d8754z-
Max-Forwards: 69
Contact: <sip:testphone@44.55.66.77:41890;transport=UDP>
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 2 ACK
Proxy-Authorization: Digest username="testphone",realm="11.22.33.44",nonce="51c8bda900011d9d670190766d75d7a751c7d9f0f4fd3393",uri="sip:+4412345@11.22.33.44;transport=UDP",response="eb6e5f32d4004e604a80be8c17349e5e",algorithm=MD5
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---

<--- SIP read from UDP:11.22.33.44:5060 --->
INVITE sip:+4412345@88.99.11.22:5060 SIP/2.0
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes>
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKc637.56117dc7.0
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-bcd7fdc83d9ec2c1-1---d8754z-
Max-Forwards: 69
Contact: <sip:testphone@44.55.66.77:41890;transport=UDP>
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest username="testphone",realm="11.22.33.44",nonce="51c8bda900011d9d670190766d75d7a751c7d9f0f4fd3393",uri="sip:+4412345@88.99.11.22:5060",response="c40a13eea34f49fad7cd778b44df4180",algorithm=MD5
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.39 r16838
Allow-Events: presence, kpml
Content-Length: 403

v=0
o=Zoiper_user 0 1 IN IP4 192.168.5.1
s=Zoiper_session
c=IN IP4 44.55.66.77
t=0 0
m=image 8000 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy
a=direction:active
a=oldmediaip:192.168.5.1
<------------->
--- (17 headers 17 lines) ---
Sending to 11.22.33.44:5060 (NAT)
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Got T.38 offer in SDP in dialog YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

<--- Transmitting (NAT) to 11.22.33.44:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKc637.56117dc7.0;received=11.22.33.44;rport=5060
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-bcd7fdc83d9ec2c1-1---d8754z-
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes>
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 3 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+4412345@88.99.11.22:5060>
Content-Length: 0


<------------>
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
set_destination: Parsing <sip:33.55.77.99:5060> for address/port to send to
set_destination: set destination to 33.55.77.99:5060
Reliably Transmitting (no NAT) to 33.55.77.99:5060:
INVITE sip:33.55.77.99:5060 SIP/2.0
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK2ea08cac
Max-Forwards: 70
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Contact: <sip:+4466666@88.99.11.22:5060>
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 103 INVITE
User-Agent: Asteriskserver
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 253

v=0
o=root 1707261853 1707261854 IN IP4 88.99.11.22
s=Asteriskserver SDP
c=IN IP4 88.99.11.22
t=0 0
m=image 42859 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:204
a=T38FaxUdpEC:t38UDPFEC

---

<--- SIP read from UDP:33.55.77.99:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK2ea08cac
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 103 INVITE
Server: Asteriskserver
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:33.55.77.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK2ea08cac
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 103 INVITE
Content-Type: application/sdp
Supported: replaces, 100rel, timer
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE
Session-Expires: 1800;refresher=uas
Require: timer
Contact: <sip:33.55.77.99:5060>
Server: Asteriskserver
Content-Length: 278

v=0
o=- 3581099020 3581099021 IN IP4 22.44.66.88
s=-
c=IN IP4 22.44.66.88
t=0 0
m=image 22122 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 12 lines) ---
Got T.38 offer in SDP in dialog 433e148975141e082e378e774d3f532c@88.99.11.22:5060
Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
set_destination: Parsing <sip:33.55.77.99:5060> for address/port to send to
set_destination: set destination to 33.55.77.99:5060
Transmitting (no NAT) to 33.55.77.99:5060:
ACK sip:33.55.77.99:5060 SIP/2.0
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK56d574cc
Max-Forwards: 70
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Contact: <sip:+4466666@88.99.11.22:5060>
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 103 ACK
User-Agent: Asteriskserver
Content-Length: 0


---

<--- Reliably Transmitting (NAT) to 11.22.33.44:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKc637.56117dc7.0;received=11.22.33.44;rport=5060
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-bcd7fdc83d9ec2c1-1---d8754z-
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes>
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 3 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+4412345@88.99.11.22:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1894142497 1894142499 IN IP4 88.99.11.22
s=Asteriskserver SDP
c=IN IP4 88.99.11.22
t=0 0
m=image 57681 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:410
a=T38FaxUdpEC:t38UDPRedundancy

<------------>

<--- SIP read from UDP:11.22.33.44:5060 --->
ACK sip:+4412345@88.99.11.22:5060 SIP/2.0
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes>
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bKc637.56117dc7.2
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-90dbf8b021152d84-1---d8754z-
Max-Forwards: 69
Contact: <sip:testphone@44.55.66.77:41890;transport=UDP>
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 3 ACK
Proxy-Authorization: Digest username="testphone",realm="11.22.33.44",nonce="51c8bda900011d9d670190766d75d7a751c7d9f0f4fd3393",uri="sip:+4412345@88.99.11.22:5060",response="c40a13eea34f49fad7cd778b44df4180",algorithm=MD5
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->

<--- SIP read from UDP:33.55.77.99:5060 --->
INVITE sip:+4466666@88.99.11.22:5060 SIP/2.0
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.1f394343.0
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
CSeq: 2 INVITE
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
Max-Forwards: 70
Content-Length: 278
User-Agent: Asteriskserver
Content-Type: application/sdp
Supported: replaces, 100rel, timer
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Contact: <sip:33.55.77.99:5060>

v=0
o=- 3581099020 3581099022 IN IP4 22.44.66.88
s=-
c=IN IP4 22.44.66.88
t=0 0
m=image 22122 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (15 headers 12 lines) ---
Sending to 33.55.77.99:5060 (no NAT)
Got T.38 offer in SDP in dialog 433e148975141e082e378e774d3f532c@88.99.11.22:5060
Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

<--- Transmitting (no NAT) to 33.55.77.99:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.1f394343.0;received=33.55.77.99
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 2 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+4466666@88.99.11.22:5060>
Content-Length: 0


<------------>

<--- Reliably Transmitting (no NAT) to 33.55.77.99:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.1f394343.0;received=33.55.77.99
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 2 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------>

<--- SIP read from UDP:33.55.77.99:5060 --->
ACK sip:+4466666@88.99.11.22:5060 SIP/2.0
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.1f394343.0
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: Asteriskserver
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:33.55.77.99:5060 --->
ACK sip:+4466666@88.99.11.22:5060 SIP/2.0
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK8074.2f394343.0
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
CSeq: 2 ACK
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
Max-Forwards: 70
Content-Length: 0
User-Agent: Asteriskserver
Contact: <sip:33.55.77.99:5060>

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:33.55.77.99:5060 --->
INVITE sip:+4466666@88.99.11.22:5060 SIP/2.0
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK9074.8221f213.0
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
CSeq: 3 INVITE
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
Max-Forwards: 70
Content-Length: 164
User-Agent: Asteriskserver
Content-Type: application/sdp
Supported: replaces, 100rel, timer
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER, UPDATE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Contact: <sip:33.55.77.99:5060>

v=0
o=- 3581099020 3581099023 IN IP4 22.44.66.88
s=-
c=IN IP4 22.44.66.88
t=0 0
m=audio 22122 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sendrecv
a=ptime:20
<------------->
--- (15 headers 9 lines) ---
Sending to 33.55.77.99:5060 (no NAT)
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 22.44.66.88:22122

<--- Transmitting (no NAT) to 33.55.77.99:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK9074.8221f213.0;received=33.55.77.99
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 3 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+4466666@88.99.11.22:5060>
Content-Length: 0


<------------>
Audio is at 17550
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 33.55.77.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK9074.8221f213.0;received=33.55.77.99
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 3 INVITE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:+4466666@88.99.11.22:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 195

v=0
o=root 1707261853 1707261855 IN IP4 88.99.11.22
s=Asteriskserver SDP
c=IN IP4 88.99.11.22
t=0 0
m=audio 17550 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:33.55.77.99:5060 --->
ACK sip:+4466666@88.99.11.22:5060 SIP/2.0
Via: SIP/2.0/UDP 33.55.77.99:5060;branch=z9hG4bK9074.9221f213.0
To: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
From: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
CSeq: 3 ACK
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
Max-Forwards: 70
Content-Length: 0
User-Agent: Asteriskserver
Contact: <sip:33.55.77.99:5060>

<------------->

<--- SIP read from UDP:11.22.33.44:5060 --->
BYE sip:+4412345@88.99.11.22:5060 SIP/2.0
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes>
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK9637.218d4292.0
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-ceb627e82ff8607e-1---d8754z-
Max-Forwards: 69
Contact: <sip:testphone@44.55.66.77:41890;transport=UDP>
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 4 BYE
Proxy-Authorization: Digest username="testphone",realm="11.22.33.44",nonce="51c8bda900011d9d670190766d75d7a751c7d9f0f4fd3393",uri="sip:+4412345@88.99.11.22:5060",response="6fdd6777c7d07c453b47cb57e88ce3ff",algorithm=MD5
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 11.22.33.44:5060 (NAT)
Scheduling destruction of SIP dialog 'YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 11.22.33.44:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK9637.218d4292.0;received=11.22.33.44;rport=5060
Via: SIP/2.0/UDP 192.168.5.1:5060;rport=41890;received=44.55.66.77;branch=z9hG4bK-d8754z-ceb627e82ff8607e-1---d8754z-
Record-Route: <sip:11.22.33.44;lr=on;ftag=7675a242;nat=yes>
From: "+4466666"<sip:in.from.a@11.22.33.44>;tag=7675a242
To: <sip:+4412345@11.22.33.44;transport=UDP>;tag=as02cf8d7c
Call-ID: YTZhMjUzYmQ2ODU3NWE2YjVkNTU3ODc0ZmNhYTQ4ODU.
CSeq: 4 BYE
Server: Asteriskserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [h@in.from.in.from.a:1] NoOp("SIP/in.from.a-00000086", "bye") in new stack
  == Spawn extension (in.from.in.from.a, h, 1) exited non-zero on 'SIP/out.to.b-00000087'
Scheduling destruction of SIP dialog '433e148975141e082e378e774d3f532c@88.99.11.22:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:33.55.77.99:5060> for address/port to send to
set_destination: set destination to 33.55.77.99:5060
Reliably Transmitting (no NAT) to 33.55.77.99:5060:
BYE sip:33.55.77.99:5060 SIP/2.0
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK55961fa5
Max-Forwards: 70
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 104 BYE
User-Agent: Asteriskserver
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (in.from.in.from.a, +4412345, 2) exited non-zero on 'SIP/in.from.a-00000086'

<--- SIP read from UDP:33.55.77.99:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.11.22:5060;branch=z9hG4bK55961fa5
From: "+4466666" <sip:+4466666@88.99.11.22>;tag=as5160c622
To: <sip:+4412345@33.55.77.99>;tag=CALL.138.211
Call-ID: 433e148975141e082e378e774d3f532c@88.99.11.22:5060
CSeq: 104 BYE
Supported: replaces, 100rel, timer
Contact: <sip:33.55.77.99:5060>
Server: Asteriskserver
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '433e148975141e082e378e774d3f532c@88.99.11.22:5060' Method: ACK