INVITE sip:101@192.168.70.129 SIP/2.

0
Via: SIP/2.0/UDP 192.168.70.1:23762;branch=z9hG4bK-d8754z-b57ee02a8e25c85f-1---
d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.70.1:23762>
To: "101"<sip:101@192.168.70.129>
From: "100"<sip:100@192.168.70.129>;tag=6a122732
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 364

v=0
o=- 6 2 IN IP4 192.168.70.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.70.1
t=0 0
m=audio 44206 RTP/AVP 107 0 8 101
a=alt:1 3 : gpm9O/uP E3VgLHXv 192.168.1.2 44206
a=alt:2 2 : 5bi1qIjn F/H8bZOM 192.168.126.1 44206
a=alt:3 1 : O4tXIsnV way+jzCj 192.168.70.1 44206
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 13 lines) ---
Sending to 192.168.70.1 : 23762 (NAT)
Using INVITE request as basis request -
MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.

<--- Reliably Transmitting (no NAT) to 192.168.70.1:23762 --->

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.70.1:23762;branch=z9hG4bK-d8754z-b57ee02a8e25c85f-1---
d8754z-
;received=192.168.70.1
;rport=23762
From: "100"<sip:100@192.168.70.129>;tag=6a122732
To: "101"<sip:101@192.168.70.129>;tag=as401c60a8
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f056cd9"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog
'MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.' in 32000 ms
(Method: INVITE)
Found user '100'

<--- SIP read from 192.168.70.1:23762 --->

ACK sip:101@192.168.70.129 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.1:23762;branch=z9hG4bK-d8754z-b57ee02a8e25c85f-1---
d8754z-
;rport
To: "101"<sip:101@192.168.70.129>;tag=as401c60a8
From: "100"<sip:100@192.168.70.129>;tag=6a122732
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from 192.168.70.1:23762 --->

INVITE sip:101@192.168.70.129 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.1:23762;branch=z9hG4bK-d8754z-db4b6a547b3a9f47-
1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.70.1:23762>
To: "101"<sip:101@192.168.70.129>
From: "100"<sip:100@192.168.70.129>;tag=6a122732
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest
username="100",realm="asterisk",nonce="1f056cd9",uri="sip:101@192.168.70.129",res
ponse="e6ee3be8c15baa045d83a3de08daa862",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 364
v=0
o=- 6 2 IN IP4 192.168.70.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.70.1
t=0 0
m=audio 44206 RTP/AVP 107 0 8 101
a=alt:1 3 : gpm9O/uP E3VgLHXv 192.168.1.2 44206
a=alt:2 2 : 5bi1qIjn F/H8bZOM 192.168.126.1 44206
a=alt:3 1 : O4tXIsnV way+jzCj 192.168.70.1 44206
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Sending to 192.168.70.1 : 23762 (NAT)
Using INVITE request as basis request -
MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
Found user '100'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.70.1:44206
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=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 192.168.70.1:44206
Looking for 101 in internal (domain 192.168.70.129)
list_route: hop: <sip:100@192.168.70.1:23762>

<--- Transmitting (no NAT) to 192.168.70.1:23762 --->

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.70.1:23762;branch=z9hG4bK-d8754z-db4b6a547b3a9f47-
1---d8754z-;received=192.168.70.1;rport=23762
From: "100"<sip:100@192.168.70.129>;tag=6a122732
To: "101"<sip:101@192.168.70.129>
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.70.129>
Content-Length: 0

<------------>
    -- Executing [101@internal:1] Dial("SIP/100-08210e28", "Sip/101") in new stack
Audio is at 192.168.70.129 port 11444
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 192.168.1.2:5060:
INVITE sip:101@192.168.1.2:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK260efc79;rport
From: "100" <sip:100@192.168.70.129>;tag=as2cad8232
To: <sip:101@192.168.1.2:5060;transport=UDP>
Contact: <sip:100@192.168.70.129>
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 25 Aug 2010 15:04:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 5582 5582 IN IP4 192.168.70.129
s=session
c=IN IP4 192.168.70.129
t=0 0
m=audio 11444 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 101
ngan-desktop*CLI>
<--- SIP read from 192.168.1.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.70.129:5060;rport=3482;received=192.168.1.2;branch=z9hG4bK260efc79
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
From: "100" <sip:100@192.168.70.129>;tag=as2cad8232
To: <sip:101@192.168.1.2>
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
ngan-desktop*CLI>
<--- SIP read from 192.168.1.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.70.129:5060;rport=3482;received=192.168.1.2;branch=z9hG4bK260efc79
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
From: "100" <sip:100@192.168.70.129>;tag=as2cad8232
To: <sip:101@192.168.1.2>;tag=3dd0b5f06a1d41b4a80a8a1bc181d06f
CSeq: 102 INVITE
Contact: <sip:192.168.1.2:5060;transport=UDP>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- SIP/101-08215ae8 is ringing

<--- Transmitting (no NAT) to 192.168.70.1:23762 --->

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.70.1:23762;branch=z9hG4bK-d8754z-db4b6a547b3a9f47-
1---d8754z-;received=192.168.70.1;rport=23762
From: "100"<sip:100@192.168.70.129>;tag=6a122732
To: "101"<sip:101@192.168.70.129>;tag=as7b40993a
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.70.129>
Content-Length: 0

<------------>
ngan-desktop*CLI>
<--- SIP read from 192.168.1.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.70.129:5060;rport=3482;received=192.168.1.2;branch=z9hG4bK260efc79
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
From: "100" <sip:100@192.168.70.129>;tag=as2cad8232
To: <sip:101@192.168.1.2>;tag=3dd0b5f06a1d41b4a80a8a1bc181d06f
CSeq: 102 INVITE
Contact: <sip:192.168.1.2:5060;transport=UDP>
Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER,
MESSAGE, OPTIONS
Supported: replaces, norefersub
Content-Type: application/sdp
Content-Length: 239

v=0
o=- 3491762680 3491762681 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 23000 RTP/AVP 0 101
a=rtcp:23001 IN IP4 192.168.1.2
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.2:23000
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing),
combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event),
combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.2:23000
list_route: hop: <sip:192.168.1.2:5060;transport=UDP>
set_destination: Parsing <sip:192.168.1.2:5060;transport=UDP> for address/port to send
to
set_destination: set destination to 192.168.1.2, port 5060
Transmitting (no NAT) to 192.168.1.2:5060:
ACK sip:192.168.1.2:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK4b9f79d7;rport
From: "100" <sip:100@192.168.70.129>;tag=as2cad8232
To:
<sip:101@192.168.1.2:5060;transport=UDP>;tag=3dd0b5f06a1d41b4a80a8a1bc181d06f
Contact: <sip:100@192.168.70.129>
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
    -- SIP/101-08215ae8 answered SIP/100-08210e28
Audio is at 192.168.70.129 port 15894
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 192.168.70.1:23762 --->

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.70.1:23762;branch=z9hG4bK-d8754z-db4b6a547b3a9f47-
1---d8754z-;received=192.168.70.1;rport=23762
From: "100"<sip:100@192.168.70.129>;tag=6a122732
To: "101"<sip:101@192.168.70.129>;tag=as7b40993a
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.70.129>
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 5582 5582 IN IP4 192.168.70.129
s=session
c=IN IP4 192.168.70.129
t=0 0
m=audio 15894 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
    -- Native bridging SIP/100-08210e28 and SIP/101-08215ae8
set_destination: Parsing <sip:192.168.1.2:5060;transport=UDP> for address/port to send
to
set_destination: set destination to 192.168.1.2, port 5060
Audio is at 192.168.70.129 port 11444
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
INVITE sip:192.168.1.2:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK704c3ce6;rport
From: "100" <sip:100@192.168.70.129>;tag=as2cad8232
To:
<sip:101@192.168.1.2:5060;transport=UDP>;tag=3dd0b5f06a1d41b4a80a8a1bc181d06f
Contact: <sip:100@192.168.70.129>
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 5582 5583 IN IP4 192.168.70.1
s=session
c=IN IP4 192.168.70.1
t=0 0
m=audio 44206 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
ngan-desktop*CLI>
<--- SIP read from 192.168.70.1:23762 --->
ACK sip:101@192.168.70.129 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.1:23762;branch=z9hG4bK-d8754z-b31688467b0ca107-
1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.70.1:23762>
To: "101"<sip:101@192.168.70.129>;tag=as7b40993a
From: "100"<sip:100@192.168.70.129>;tag=6a122732
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 2 ACK
Proxy-Authorization: Digest
username="100",realm="asterisk",nonce="1f056cd9",uri="sip:101@192.168.70.129",res
ponse="e6ee3be8c15baa045d83a3de08daa862",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:100@192.168.70.1:23762> for address/port to send to
set_destination: set destination to 192.168.70.1, port 23762
Audio is at 192.168.70.129 port 15894
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.70.1:23762:
INVITE sip:100@192.168.70.1:23762 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK2f4d62bb;rport
From: "101"<sip:101@192.168.70.129>;tag=as7b40993a
To: "100"<sip:100@192.168.70.129>;tag=6a122732
Contact: <sip:101@192.168.70.129>
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 5582 5583 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 23000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
ngan-desktop*CLI>
<--- SIP read from 192.168.70.1:23762 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK2f4d62bb;rport=5060
Contact: <sip:100@192.168.70.1:23762>
To: "100"<sip:100@192.168.70.129>;tag=6a122732
From: "101"<sip:101@192.168.70.129>;tag=as7b40993a
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 6 3 IN IP4 192.168.70.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.70.1
t=0 0
m=audio 44206 RTP/AVP 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.70.1:44206
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined -
0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event),
combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.70.1:44206
set_destination: Parsing <sip:100@192.168.70.1:23762> for address/port to send to
set_destination: set destination to 192.168.70.1, port 23762
Transmitting (no NAT) to 192.168.70.1:23762:
ACK sip:100@192.168.70.1:23762 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK6cb757d8;rport
From: "101"<sip:101@192.168.70.129>;tag=as7b40993a
To: "100"<sip:100@192.168.70.129>;tag=6a122732
Contact: <sip:101@192.168.70.129>
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
ngan-desktop*CLI>
<--- SIP read from 192.168.1.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.70.129:5060;rport=3482;received=192.168.1.2;branch=z9hG4bK704c3ce6
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
From: "100" <sip:100@192.168.70.129>;tag=as2cad8232
To: <sip:101@192.168.1.2>;tag=3dd0b5f06a1d41b4a80a8a1bc181d06f
CSeq: 103 INVITE
Contact: <sip:192.168.1.2:5060;transport=UDP>
Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER,
MESSAGE, OPTIONS
Supported: replaces, norefersub
Content-Type: application/sdp
Content-Length: 239

v=0
o=- 3491762683 3491762682 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 23000 RTP/AVP 0 101
a=rtcp:23001 IN IP4 192.168.1.2
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.2:23000
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing),
combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event),
combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.2:23000
set_destination: Parsing <sip:192.168.1.2:5060;transport=UDP> for address/port to send
to
set_destination: set destination to 192.168.1.2, port 5060
Transmitting (no NAT) to 192.168.1.2:5060:
ACK sip:192.168.1.2:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK41ab1bfe;rport
From: "100" <sip:100@192.168.70.129>;tag=as2cad8232
To:
<sip:101@192.168.1.2:5060;transport=UDP>;tag=3dd0b5f06a1d41b4a80a8a1bc181d06f
Contact: <sip:100@192.168.70.129>
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
ngan-desktop*CLI>
<--- SIP read from 192.168.70.1:5060 --->
BYE sip:100@192.168.70.129 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.2:5060;rport;branch=z9hG4bKPj04eed753bf4e4a3ab357ca325d6c0278
Max-Forwards: 70
From: <sip:101@192.168.1.2>;tag=3dd0b5f06a1d41b4a80a8a1bc181d06f
To: "100" <sip:100@192.168.70.129>;tag=as2cad8232
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
CSeq: 1674275836 BYE
User-Agent: VidoSIP/1.5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.70.1 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.70.1:5060 --->

SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bKPj04eed753bf4e4a3ab357ca325d6c0278;received=19
2.168.70.1;rport=5060
From: <sip:101@192.168.1.2>;tag=3dd0b5f06a1d41b4a80a8a1bc181d06f
To: "100" <sip:100@192.168.70.129>;tag=as2cad8232
Call-ID: 15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129
CSeq: 1674275836 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@192.168.70.129>
Content-Length: 0

<------------>
set_destination: Parsing <sip:100@192.168.70.1:23762> for address/port to send to
set_destination: set destination to 192.168.70.1, port 23762
Audio is at 192.168.70.129 port 15894
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.70.1:23762:
INVITE sip:100@192.168.70.1:23762 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK2a7461d3;rport
From: "101"<sip:101@192.168.70.129>;tag=as7b40993a
To: "100"<sip:100@192.168.70.129>;tag=6a122732
Contact: <sip:101@192.168.70.129>
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 5582 5584 IN IP4 192.168.70.129
s=session
c=IN IP4 192.168.70.129
t=0 0
m=audio 15894 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
  == Spawn extension (internal, 101, 1) exited non-zero on 'SIP/100-08210e28'
Scheduling destruction of SIP dialog
'MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.' in 32000 ms
(Method: ACK)

<--- SIP read from 192.168.70.1:5060 --->

REGISTER sip:192.168.70.129:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.2:5060;rport;branch=z9hG4bKPjb0b2109116e94409b6724d1e2b7d9d61
Max-Forwards: 70
From: <sip:101@192.168.70.129>;tag=203c01ed87fd450b998ab892d7e5ddb9
To: <sip:101@192.168.70.129>
Call-ID: a052a313dd01423bacf93e7054fe12a8
CSeq: 18 REGISTER
User-Agent: VidoSIP/1.5
Contact: <sip:101@192.168.1.2:5060;transport=UDP>;methods="INVITE, INFO,
OPTIONS, BYE, CANCEL, ACK"
Expires: 600
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.70.1 : 5060 (NAT)

<--- Transmitting (no NAT) to 192.168.1.2:5060 --->

SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bKPjb0b2109116e94409b6724d1e2b7d9d61;received=1
92.168.70.1;rport=5060
From: <sip:101@192.168.70.129>;tag=203c01ed87fd450b998ab892d7e5ddb9
To: <sip:101@192.168.70.129>
Call-ID: a052a313dd01423bacf93e7054fe12a8
CSeq: 18 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.70.129>
Content-Length: 0

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

<--- Transmitting (no NAT) to 192.168.1.2:5060 --->

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bKPjb0b2109116e94409b6724d1e2b7d9d61;received=1
92.168.70.1;rport=5060
From: <sip:101@192.168.70.129>;tag=203c01ed87fd450b998ab892d7e5ddb9
To: <sip:101@192.168.70.129>;tag=as0a628145
Call-ID: a052a313dd01423bacf93e7054fe12a8
CSeq: 18 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22844f1e"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'a052a313dd01423bacf93e7054fe12a8' in 32000 ms
(Method: REGISTER)
Really destroying SIP dialog '15b5cdea5b7a8c2256f761672f1dc319@192.168.70.129'
Method: BYE

<--- SIP read from 192.168.70.1:5060 --->

REGISTER sip:192.168.70.129:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.2:5060;rport;branch=z9hG4bKPj380b5eaa086348d4a683e8bfbf90c75c
Max-Forwards: 70
From: <sip:101@192.168.70.129>;tag=203c01ed87fd450b998ab892d7e5ddb9
To: <sip:101@192.168.70.129>
Call-ID: a052a313dd01423bacf93e7054fe12a8
CSeq: 19 REGISTER
User-Agent: VidoSIP/1.5
Contact: <sip:101@192.168.1.2:5060;transport=UDP>;methods="INVITE, INFO,
OPTIONS, BYE, CANCEL, ACK"
Expires: 600
Authorization: Digest username="101", realm="asterisk", nonce="22844f1e",
uri="sip:192.168.70.129:5060", response="689bb0c816c6ffa9e2d4dc2e9917a61a",
algorithm=md5
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.70.1 : 5060 (NAT)

<--- Transmitting (no NAT) to 192.168.1.2:5060 --->

SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bKPj380b5eaa086348d4a683e8bfbf90c75c;received=19
2.168.70.1;rport=5060
From: <sip:101@192.168.70.129>;tag=203c01ed87fd450b998ab892d7e5ddb9
To: <sip:101@192.168.70.129>
Call-ID: a052a313dd01423bacf93e7054fe12a8
CSeq: 19 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@192.168.70.129>
Content-Length: 0

<------------>
ngan-desktop*CLI>
<--- Transmitting (no NAT) to 192.168.1.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bKPj380b5eaa086348d4a683e8bfbf90c75c;received=19
2.168.70.1;rport=5060
From: <sip:101@192.168.70.129>;tag=203c01ed87fd450b998ab892d7e5ddb9
To: <sip:101@192.168.70.129>;tag=as0a628145
Call-ID: a052a313dd01423bacf93e7054fe12a8
CSeq: 19 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 600
Contact: <sip:101@192.168.1.2:5060;transport=UDP>;expires=600
Date: Wed, 25 Aug 2010 15:04:53 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'a052a313dd01423bacf93e7054fe12a8' in 32000 ms
(Method: REGISTER)
ngan-desktop*CLI>
<--- SIP read from 192.168.70.1:23762 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK2a7461d3;rport=5060
Contact: <sip:100@192.168.70.1:23762>
To: "100"<sip:100@192.168.70.129>;tag=6a122732
From: "101"<sip:101@192.168.70.129>;tag=as7b40993a
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 6 3 IN IP4 192.168.70.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.70.1
t=0 0
m=audio 44206 RTP/AVP 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.70.1:44206
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined -
0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event),
combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.70.1:44206
set_destination: Parsing <sip:100@192.168.70.1:23762> for address/port to send to
set_destination: set destination to 192.168.70.1, port 23762
Transmitting (no NAT) to 192.168.70.1:23762:
ACK sip:100@192.168.70.1:23762 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK11a6317c;rport
From: "101"<sip:101@192.168.70.129>;tag=as7b40993a
To: "100"<sip:100@192.168.70.129>;tag=6a122732
Contact: <sip:101@192.168.70.129>
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
set_destination: Parsing <sip:100@192.168.70.1:23762> for address/port to send to
set_destination: set destination to 192.168.70.1, port 23762
Reliably Transmitting (no NAT) to 192.168.70.1:23762:
BYE sip:100@192.168.70.1:23762 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.129:5060;branch=z9hG4bK752a72b1;rport
From: "101"<sip:101@192.168.70.129>;tag=as7b40993a
To: "100"<sip:100@192.168.70.129>;tag=6a122732
Call-ID: MTM0NDJlNzFiZTY4YTVhN2YwMGMzOWI5OGUzY2I4NmQ.
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

;;;;;;;;;;;;;;;;;;;;;;;HAI SERVER ASTERISK QUA SIP TRUNK;;;;;;;;;;;;;;;;;;;;;;;

cisco-desktop*CLI> sip debug

SIP Debugging enabled
cisco-desktop*CLI>
<--- SIP read from 10.215.23.17:57496 --->
INVITE sip:1300@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.17:57496;branch=z9hG4bK-d8754z-7d53bd60506ca822-
1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1400@10.215.23.17:57496>
To: "1300"<sip:1300@10.215.23.64>
From: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 316

v=0
o=- 3 2 IN IP4 10.215.23.17
s=CounterPath X-Lite 3.0
c=IN IP4 10.215.23.17
t=0 0
m=audio 22632 RTP/AVP 107 0 8 101
a=alt:1 2 : WhRW6sAP qaIGPajZ 169.254.25.129 22632
a=alt:2 1 : 4L7JjQtA UoxzRXvt 10.215.23.17 22632
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
Sending to 10.215.23.17 : 57496 (NAT)
Using INVITE request as basis request -
MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.

<--- Reliably Transmitting (no NAT) to 10.215.23.17:57496 --->

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.215.23.17:57496;branch=z9hG4bK-d8754z-7d53bd60506ca822-
1---d8754z-;received=10.215.23.17;rport=57496
From: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
To: "1300"<sip:1300@10.215.23.64>;tag=as2f9ed494
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="546cd5fc"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog
'MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.' in 32000 ms (Method:
INVITE)
Found user '1400'
cisco-desktop*CLI>
<--- SIP read from 10.215.23.17:57496 --->
ACK sip:1300@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.17:57496;branch=z9hG4bK-d8754z-7d53bd60506ca822-
1---d8754z-;rport
To: "1300"<sip:1300@10.215.23.64>;tag=as2f9ed494
From: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
cisco-desktop*CLI>
<--- SIP read from 10.215.23.17:57496 --->
INVITE sip:1300@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.17:57496;branch=z9hG4bK-d8754z-175fe648c21ea63a-1---
d8754z-;rport
Max-Forwards: 70
Contact: <sip:1400@10.215.23.17:57496>
To: "1300"<sip:1300@10.215.23.64>
From: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest
username="1400",realm="asterisk",nonce="546cd5fc",uri="sip:1300@10.215.23.64",res
ponse="83f3a1c6715139fcda3b573cb130c180",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 316

v=0
o=- 3 2 IN IP4 10.215.23.17
s=CounterPath X-Lite 3.0
c=IN IP4 10.215.23.17
t=0 0
m=audio 22632 RTP/AVP 107 0 8 101
a=alt:1 2 : WhRW6sAP qaIGPajZ 169.254.25.129 22632
a=alt:2 1 : 4L7JjQtA UoxzRXvt 10.215.23.17 22632
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 12 lines) ---
Sending to 10.215.23.17 : 57496 (NAT)
Using INVITE request as basis request -
MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
Found user '1400'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.215.23.17:22632
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|
alaw)/video=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 10.215.23.17:22632
Looking for 1300 in outbound (domain 10.215.23.64)
list_route: hop: <sip:1400@10.215.23.17:57496>

<--- Transmitting (no NAT) to 10.215.23.17:57496 --->

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.215.23.17:57496;branch=z9hG4bK-d8754z-175fe648c21ea63a-1---
d8754z-;received=10.215.23.17;rport=57496
From: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
To: "1300"<sip:1300@10.215.23.64>
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.64>
Content-Length: 0

<------------>
    -- Executing [1300@outbound:1] Dial("SIP/1400-081e65b0", "SIP/1300@server2") in
new stack
Audio is at 10.215.23.64 port 17938
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.215.23.63:5060:
INVITE sip:1300@10.215.23.63 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK46763507;rport
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>
Contact: <sip:1400@10.215.23.64>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Aug 2010 10:49:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 6230 6230 IN IP4 10.215.23.64
s=session
c=IN IP4 10.215.23.64
t=0 0
m=audio 17938 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecvtop*CLI>

---
    -- Called 1300@server2
cisco-desktop*CLI>
<--- SIP read from 10.215.23.63:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.215.23.64:5060;branch=z9hG4bK46763507;received=10.215.23.64;rport=5060
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.63>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
cisco-desktop*CLI>
<--- SIP read from 10.215.23.63:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.215.23.64:5060;branch=z9hG4bK46763507;received=10.215.23.64;rport=5060
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.63>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- SIP/server2-081f48a8 is ringing
<--- Transmitting (no NAT) to 10.215.23.17:57496 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.215.23.17:57496;branch=z9hG4bK-d8754z-175fe648c21ea63a-1---
d8754z-;received=10.215.23.17;rport=57496
From: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
To: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.64>
Content-Length: 0

<------------>
cisco-desktop*CLI>
<--- SIP read from 10.215.23.63:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.215.23.64:5060;branch=z9hG4bK46763507;received=10.215.23.64;rport=5060
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.63>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 6982 6982 IN IP4 10.215.23.63
s=session
c=IN IP4 10.215.23.63
t=0 0
m=audio 13454 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 14 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.215.23.63:13454
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|
alaw)/video=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 10.215.23.63:13454
list_route: hop: <sip:1300@10.215.23.63>
set_destination: Parsing <sip:1300@10.215.23.63> for address/port to send to
set_destination: set destination to 10.215.23.63, port 5060
Transmitting (no NAT) to 10.215.23.63:5060:
ACK sip:1300@10.215.23.63 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK2936e683;rport
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Contact: <sip:1400@10.215.23.64>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
    -- SIP/server2-081f48a8 answered SIP/1400-081e65b0
Audio is at 10.215.23.64 port 10356
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 10.215.23.17:57496 --->

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.215.23.17:57496;branch=z9hG4bK-d8754z-175fe648c21ea63a-1---
d8754z-;received=10.215.23.17;rport=57496
From: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
To: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.64>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6230 6230 IN IP4 10.215.23.64
s=session
c=IN IP4 10.215.23.64
t=0 0
m=audio 10356 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Native bridging SIP/1400-081e65b0 and SIP/server2-081f48a8
set_destination: Parsing <sip:1300@10.215.23.63> for address/port to send to
set_destination: set destination to 10.215.23.63, port 5060
Audio is at 10.215.23.64 port 17938
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 10.215.23.63:5060:
INVITE sip:1300@10.215.23.63 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK12ad2640;rport
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Contact: <sip:1400@10.215.23.64>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6230 6231 IN IP4 10.215.23.17
s=session
c=IN IP4 10.215.23.17
t=0 0
m=audio 22632 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from 10.215.23.63:5060 --->

INVITE sip:1400@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.63:5060;branch=z9hG4bK07e14de0;rport
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Contact: <sip:1300@10.215.23.63>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 6982 6983 IN IP4 10.215.23.101
s=session
c=IN IP4 10.215.23.101
t=0 0
m=audio 18486 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---

<--- Transmitting (no NAT) to 10.215.23.63:5060 --->

SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP
10.215.23.63:5060;branch=z9hG4bK07e14de0;received=10.215.23.63;rport=5060
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

<------------>
cisco-desktop*CLI>
<--- SIP read from 10.215.23.63:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP
10.215.23.64:5060;branch=z9hG4bK12ad2640;received=10.215.23.64;rport=5060
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
-Asterisk-HangupCauseCode: 16

<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:1300@10.215.23.63> for address/port to send to
set_destination: set destination to 10.215.23.63, port 5060
Transmitting (no NAT) to 10.215.23.63:5060:
ACK sip:1300@10.215.23.63 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK12ad2640;rport
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Contact: <sip:1400@10.215.23.64>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
cisco-desktop*CLI>
<--- SIP read from 10.215.23.63:5060 --->
ACK sip:1400@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.63:5060;branch=z9hG4bK07e14de0;rport
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Contact: <sip:1300@10.215.23.63>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
cisco-desktop*CLI>
<--- SIP read from 10.215.23.17:57496 --->
ACK sip:1300@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.17:57496;branch=z9hG4bK-d8754z-c978a5233205636e-
1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1400@10.215.23.17:57496>
To: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
From: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 2 ACK
Proxy-Authorization: Digest
username="1400",realm="asterisk",nonce="546cd5fc",uri="sip:1300@10.215.23.64",res
ponse="83f3a1c6715139fcda3b573cb130c180",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:1400@10.215.23.17:57496> for address/port to send to
set_destination: set destination to 10.215.23.17, port 57496
Audio is at 10.215.23.64 port 10356
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 10.215.23.17:57496:
INVITE sip:1400@10.215.23.17:57496 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK0369275b;rport
From: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
To: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Contact: <sip:1300@10.215.23.64>
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6230 6231 IN IP4 10.215.23.63
s=session
c=IN IP4 10.215.23.63
t=0 0
m=audio 13454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
cisco-desktop*CLI>
<--- SIP read from 10.215.23.17:57496 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK0369275b;rport=5060
Contact: <sip:1400@10.215.23.17:57496>
To: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
From: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 185

v=0
o=- 3 3 IN IP4 10.215.23.17
s=CounterPath X-Lite 3.0
c=IN IP4 10.215.23.17
t=0 0
m=audio 22632 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.215.23.17:22632
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=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 10.215.23.17:22632
list_route: hop: <sip:1400@10.215.23.17:57496>
set_destination: Parsing <sip:1400@10.215.23.17:57496> for address/port to send to
set_destination: set destination to 10.215.23.17, port 57496
Transmitting (no NAT) to 10.215.23.17:57496:
ACK sip:1400@10.215.23.17:57496 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK23407376;rport
From: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
To: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Contact: <sip:1300@10.215.23.64>
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
cisco-desktop*CLI>
<--- SIP read from 10.215.23.63:5060 --->
INVITE sip:1400@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.63:5060;branch=z9hG4bK45ed8ad0;rport
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Contact: <sip:1300@10.215.23.63>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6982 6984 IN IP4 10.215.23.63
s=session
c=IN IP4 10.215.23.63
t=0 0
m=audio 13454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Sending to 10.215.23.63 : 5060 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.215.23.63:13454
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=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 10.215.23.63:13454
<--- Transmitting (NAT) to 10.215.23.63:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.215.23.63:5060;branch=z9hG4bK45ed8ad0;received=10.215.23.63;rport=5060
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1400@10.215.23.64>
Content-Length: 0

<------------>
Audio is at 10.215.23.64 port 17938
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 10.215.23.63:5060 --->

SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.215.23.63:5060;branch=z9hG4bK45ed8ad0;received=10.215.23.63;rport=5060
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1400@10.215.23.64>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6230 6232 IN IP4 10.215.23.17
s=session
c=IN IP4 10.215.23.17
t=0 0
m=audio 22632 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
cisco-desktop*CLI>
<--- SIP read from 10.215.23.63:5060 --->
ACK sip:1400@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.63:5060;branch=z9hG4bK059188e6;rport
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Contact: <sip:1300@10.215.23.63>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:1300@10.215.23.63> for address/port to send to
set_destination: set destination to 10.215.23.63, port 5060
Audio is at 10.215.23.64 port 17938
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.215.23.63:5060:
INVITE sip:1300@10.215.23.63 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK5e10af3b;rport
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Contact: <sip:1400@10.215.23.64>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6230 6233 IN IP4 10.215.23.17
s=session
c=IN IP4 10.215.23.17
t=0 0
m=audio 22632 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from 10.215.23.63:5060 --->

BYE sip:1400@10.215.23.64 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.63:5060;branch=z9hG4bK4a46f95b;rport
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 10.215.23.63 : 5060 (NAT)

<--- Transmitting (NAT) to 10.215.23.63:5060 --->

SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.215.23.63:5060;branch=z9hG4bK4a46f95b;received=10.215.23.63;rport=5060
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 104 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1400@10.215.23.64>
Content-Length: 0

<------------>
set_destination: Parsing <sip:1400@10.215.23.17:57496> for address/port to send to
set_destination: set destination to 10.215.23.17, port 57496
Audio is at 10.215.23.64 port 10356
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 10.215.23.17:57496:
INVITE sip:1400@10.215.23.17:57496 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK28d8cc41;rport
From: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
To: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Contact: <sip:1300@10.215.23.64>
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6230 6232 IN IP4 10.215.23.64
s=session
c=IN IP4 10.215.23.64
t=0 0
m=audio 10356 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
  == Spawn extension (outbound, 1300, 1) exited non-zero on 'SIP/1400-081e65b0'
Scheduling destruction of SIP dialog
'MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.' in 32000 ms (Method:
ACK)
cisco-desktop*CLI>
<--- SIP read from 10.215.23.63:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP
10.215.23.64:5060;branch=z9hG4bK5e10af3b;received=10.215.23.64;rport=5060
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.63>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Got SIP response 503 "Unavailable" back from 10.215.23.63
set_destination: Parsing <sip:1300@10.215.23.63> for address/port to send to
set_destination: set destination to 10.215.23.63, port 5060
Transmitting (NAT) to 10.215.23.63:5060:
ACK sip:1300@10.215.23.63 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK5e10af3b;rport
From: <sip:1300@10.215.23.63>;tag=as596ccb25
To: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
Contact: <sip:1400@10.215.23.64>
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
Really destroying SIP dialog '2a1431133ea6f59017584f06468d7176@10.215.23.64'
Method: BYE
cisco-desktop*CLI> sip debug
<--- SIP read from 10.215.23.17:57496 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK28d8cc41;rport=5060
Contact: <sip:1400@10.215.23.17:57496>
To: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
From: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 185
v=0
o=- 3 3 IN IP4 10.215.23.17
s=CounterPath X-Lite 3.0
c=IN IP4 10.215.23.17
t=0 0
m=audio 22632 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.215.23.17:22632
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=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 10.215.23.17:22632
set_destination: Parsing <sip:1400@10.215.23.17:57496> for address/port to send to
set_destination: set destination to 10.215.23.17, port 57496
Transmitting (no NAT) to 10.215.23.17:57496:
ACK sip:1400@10.215.23.17:57496 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK1441efe8;rport
From: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
To: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Contact: <sip:1300@10.215.23.64>
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
set_destination: Parsing <sip:1400@10.215.23.17:57496> for address/port to send to
set_destination: set destination to 10.215.23.17, port 57496
Reliably Transmitting (no NAT) to 10.215.23.17:57496:
BYE sip:1400@10.215.23.17:57496 SIP/2.0
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK34fdbfc1;rport
From: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
To: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
Scheduling destruction of SIP dialog
'MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.' in 32000 ms (Method:
ACK)
cisco-desktop*CLI> sip debug
<--- SIP read from 10.215.23.17:57496 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.215.23.64:5060;branch=z9hG4bK34fdbfc1;rport=5060
Contact: <sip:1400@10.215.23.17:57496>
To: "1400"<sip:1400@10.215.23.64>;tag=b20a2e3a
From: "1300"<sip:1300@10.215.23.64>;tag=as066e2a98
Call-ID: MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.
CSeq: 104 BYE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog
'MDg2MjZhNGE0MDIwZTZjNjZkNDBhNjgxZTYyNjA1YmY.' Method: ACK
cisco-desktop*CLI> sip no debug
<--- SIP read from 10.215.23.63:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP
10.215.23.64:5060;branch=z9hG4bK5e10af3b;received=10.215.23.64;rport=5060
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.63>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
cisco-desktop*CLI> sip no debug
<--- SIP read from 10.215.23.63:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP
10.215.23.64:5060;branch=z9hG4bK5e10af3b;received=10.215.23.64;rport=5060
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.63>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
cisco-desktop*CLI> sip no debug
<--- SIP read from 10.215.23.63:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP
10.215.23.64:5060;branch=z9hG4bK5e10af3b;received=10.215.23.64;rport=5060
From: "1400" <sip:1400@10.215.23.64>;tag=as19af4dff
To: <sip:1300@10.215.23.63>;tag=as596ccb25
Call-ID: 2a1431133ea6f59017584f06468d7176@10.215.23.64
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1300@10.215.23.63>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
cisco-desktop*CLI> sip no debug

cisco-desktop*CLI> sip show peers

Name/username              Host            Dyn Nat ACL Port     Status               
1400/1400                  10.215.23.17     D          57496    Unmonitored           
server2                    10.215.23.63                5060     Unmonitored           
1000/1000                  (Unspecified)    D          0        Unmonitored           
toronto                    10.215.23.63                5060     Unmonitored