<--- SIP read from UDP:172.23.39.133:5060 --->
INVITE sip:2302@172.23.25.230 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.133:5060;branch=z9hG4bK1965889045
From: <sip:2303@172.23.25.230:5060;user=phone>;tag=676452684
To: <sip:2302@172.23.25.230>
Call-ID: 144151453@172.23.39.133
CSeq: 1 INVITE
Contact: <sip:2303@172.23.39.133:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, PRACK, REFER, NOTIFY, INFO
Content-Encoding: identity
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 151

=0
o=2303 4984 4984 IN IP4 172.23.39.133
s=-
c=IN IP4 172.23.39.133
t=0 0
m=audio 50256 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
<------------->
--- (13 headers 9 lines) ---
Sending to 172.23.39.133:5060 (no NAT)
Using INVITE request as basis request - 144151453@172.23.39.133
Found peer '2303' for '2303' from 172.23.39.133:5060

<--- Reliably Transmitting (no NAT) to 172.23.39.133:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.23.39.133:5060;branch=z9hG4bK1965889045;received=172.23.39.133
From: <sip:2303@172.23.25.230:5060;user=phone>;tag=676452684
To: <sip:2302@172.23.25.230>;tag=as189b8feb
Call-ID: 144151453@172.23.39.133
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06df418e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '144151453@172.23.39.133' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.23.39.133:5060 --->
ACK sip:2302@172.23.25.230 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.133:5060;branch=z9hG4bK1965889045
From: <sip:2303@172.23.25.230:5060;user=phone>;tag=676452684
To: <sip:2302@172.23.25.230>;tag=as189b8feb
Call-ID: 144151453@172.23.39.133
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:172.23.39.133:5060 --->
INVITE sip:2302@172.23.25.230 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.133:5060;branch=z9hG4bK584695034
From: <sip:2303@172.23.25.230:5060;user=phone>;tag=676452684
To: <sip:2302@172.23.25.230>
Call-ID: 144151453@172.23.39.133
CSeq: 2 INVITE
Contact: <sip:2303@172.23.39.133:5060>
Authorization: Digest username="2303", realm="asterisk", nonce="06df418e", uri="sip:2302@172.23.25.230", response="92cad228ad5f25fb45ca92ba2574c53b", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, PRACK, REFER, NOTIFY, INFO
Content-Encoding: identity
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 151

v=0
o=2303 4984 4984 IN IP4 172.23.39.133
s=-
c=IN IP4 172.23.39.133
t=0 0
m=audio 50256 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
<------------->
--- (14 headers 9 lines) ---
Sending to 172.23.39.133:5060 (no NAT)
Using INVITE request as basis request - 144151453@172.23.39.133
Found peer '2303' for '2303' from 172.23.39.133:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 172.23.39.133:50256
Looking for 2302 in milktest (domain 172.23.25.230)
list_route: hop: <sip:2303@172.23.39.133:5060>

<--- Transmitting (no NAT) to 172.23.39.133:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.23.39.133:5060;branch=z9hG4bK584695034;received=172.23.39.133
From: <sip:2303@172.23.25.230:5060;user=phone>;tag=676452684
To: <sip:2302@172.23.25.230>
Call-ID: 144151453@172.23.39.133
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2302@172.23.25.230:5060>
Content-Length: 0


<------------>
    -- Executing [2302@milktest:1] Dial("SIP/2303-00000014", "SIP/2302,60,Tt") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 172.23.39.108:5060:
INVITE sip:2302@172.23.39.108:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK2115e992
Max-Forwards: 70
From: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
To: <sip:2302@172.23.39.108:5060>
Contact: <sip:2303@172.23.25.230:5060>
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.0
Date: Mon, 29 Nov 2010 11:39:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 1217526626 1217526626 IN IP4 172.23.25.230
s=Asterisk PBX 1.8.0
c=IN IP4 172.23.25.230
t=0 0
m=audio 18436 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 2302

<--- SIP read from UDP:172.23.39.108:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK2115e992
From: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
To: <sip:2302@172.23.39.108:5060>
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: ZyXEL V500-Series
Content-Length: 0

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

<--- SIP read from UDP:172.23.39.108:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK2115e992
From: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
To: <sip:2302@172.23.39.108:5060>;tag=790751827
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: ZyXEL V500-Series
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- SIP/2302-00000015 is ringing

<--- Transmitting (no NAT) to 172.23.39.133:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.23.39.133:5060;branch=z9hG4bK584695034;received=172.23.39.133
From: <sip:2303@172.23.25.230:5060;user=phone>;tag=676452684
To: <sip:2302@172.23.25.230>;tag=as3e4aa344
Call-ID: 144151453@172.23.39.133
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2302@172.23.25.230:5060>
Content-Length: 0


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

<--- SIP read from UDP:172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK2115e992
From: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
To: <sip:2302@172.23.39.108:5060>;tag=790751827
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 102 INVITE
Contact: <sip:2302@172.23.39.108:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, PRACK, REFER, NOTIFY, INFO
User-Agent: ZyXEL V500-Series
Supported: timer100rel
Content-Length: 168

v=0
o=2302 8421 8421 IN IP4 172.23.39.108
s=Asterisk PBX 1.8.0
c=IN IP4 172.23.39.108
t=0 0
m=audio 50000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 172.23.39.108:50000
list_route: hop: <sip:2302@172.23.39.108:5060>
set_destination: Parsing <sip:2302@172.23.39.108:5060> for address/port to send to
set_destination: set destination to 172.23.39.108:5060
Transmitting (no NAT) to 172.23.39.108:5060:
ACK sip:2302@172.23.39.108:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK152c65f7
Max-Forwards: 70
From: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
To: <sip:2302@172.23.39.108:5060>;tag=790751827
Contact: <sip:2303@172.23.25.230:5060>
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.0
Content-Length: 0


---
    -- SIP/2302-00000015 answered SIP/2303-00000014
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 172.23.39.133:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.133:5060;branch=z9hG4bK584695034;received=172.23.39.133
From: <sip:2303@172.23.25.230:5060;user=phone>;tag=676452684
To: <sip:2302@172.23.25.230>;tag=as3e4aa344
Call-ID: 144151453@172.23.39.133
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2302@172.23.25.230:5060>
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 1322144526 1322144526 IN IP4 172.23.25.230
s=Asterisk PBX 1.8.0
c=IN IP4 172.23.25.230
t=0 0
m=audio 10462 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Locally bridging SIP/2303-00000014 and SIP/2302-00000015

<--- SIP read from UDP:172.23.39.133:5060 --->
ACK sip:2302@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.133:5060;branch=z9hG4bK1857971538
From: <sip:2303@172.23.25.230:5060;user=phone>;tag=676452684
To: <sip:2302@172.23.25.230>;tag=as3e4aa344
Call-ID: 144151453@172.23.39.133
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 0

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

<--- SIP read from UDP:172.23.39.108:5060 --->
INFO sip:2303@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1625639102
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 1 INFO
Content-Type: application/dtmf-relay
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 24

Signal=*
Duration=250
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: *

<--- Transmitting (no NAT) to 172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1625639102;received=172.23.39.108
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 1 INFO
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Locally bridging SIP/2303-00000014 and SIP/2302-00000015

<--- SIP read from UDP:172.23.39.108:5060 --->
INFO sip:2303@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1321373734
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 2 INFO
Content-Type: application/dtmf-relay
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 24

Signal=9
Duration=250
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 9

<--- Transmitting (no NAT) to 172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1321373734;received=172.23.39.108
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 2 INFO
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Locally bridging SIP/2303-00000014 and SIP/2302-00000015

<--- SIP read from UDP:172.23.39.108:5060 --->
INFO sip:2303@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK288655553
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 3 INFO
Content-Type: application/dtmf-relay
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 24

Signal=6
Duration=250
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 6

<--- Transmitting (no NAT) to 172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK288655553;received=172.23.39.108
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 3 INFO
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Started music on hold, class 'default', on SIP/2303-00000014
    -- <SIP/2302-00000015> Playing 'pbx-transfer.ulaw' (language 'en')

<--- SIP read from UDP:172.23.39.108:5060 --->
INFO sip:2303@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK279533180
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 4 INFO
Content-Type: application/dtmf-relay
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 24

Signal=2
Duration=250
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 2

<--- Transmitting (no NAT) to 172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK279533180;received=172.23.39.108
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 4 INFO
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:172.23.39.108:5060 --->
INFO sip:2303@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1148701830
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 5 INFO
Content-Type: application/dtmf-relay
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 24

Signal=3
Duration=250
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 3

<--- Transmitting (no NAT) to 172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1148701830;received=172.23.39.108
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 5 INFO
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:172.23.39.108:5060 --->
INFO sip:2303@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1062819194
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 6 INFO
Content-Type: application/dtmf-relay
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 24

Signal=0
Duration=250
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 0

<--- Transmitting (no NAT) to 172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1062819194;received=172.23.39.108
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 6 INFO
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:172.23.39.108:5060 --->
INFO sip:2303@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1738637670
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 7 INFO
Content-Type: application/dtmf-relay
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 24

Signal=1
Duration=250
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received: 1

<--- Transmitting (no NAT) to 172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK1738637670;received=172.23.39.108
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 7 INFO
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [2301@milktest:1] Dial("Local/2301@milktest-1e6d;2", "SIP/2301,60,Tt") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 172.23.49.101:16398:
INVITE sip:2301@172.23.49.101:16398;rinstance=0bec4a3d638cd5f0 SIP/2.0
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK415be79b
Max-Forwards: 70
From: "2302" <sip:2302@172.23.25.230>;tag=as5da1659a
To: <sip:2301@172.23.49.101:16398;rinstance=0bec4a3d638cd5f0>
Contact: <sip:2302@172.23.25.230:5060>
Call-ID: 40a9cc6d5cf722685ee961e10dd83ab3@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.0
Date: Mon, 29 Nov 2010 11:39:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 755113 755113 IN IP4 172.23.25.230
s=Asterisk PBX 1.8.0
c=IN IP4 172.23.25.230
t=0 0
m=audio 11400 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 2301

<--- SIP read from UDP:172.23.49.101:16398 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK415be79b
Contact: <sip:2301@172.23.49.101:16398;rinstance=0bec4a3d638cd5f0>
To: <sip:2301@172.23.49.101:16398;rinstance=0bec4a3d638cd5f0>;tag=4b21e62c
From: "2302"<sip:2302@172.23.25.230>;tag=as5da1659a
Call-ID: 40a9cc6d5cf722685ee961e10dd83ab3@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: eyeBeam release 1003l stamp 30936
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/2301-00000016 is ringing
    -- Local/2301@milktest-1e6d;1 is ringing

<--- SIP read from UDP:172.23.39.108:5060 --->
BYE sip:2303@172.23.25.230:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK2135716130
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 8 BYE
Max-Forwards: 70
User-Agent: ZyXEL V500-Series
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.23.39.108:5060 (no NAT)

Scheduling destruction of SIP dialog '0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.39.108:5060;branch=z9hG4bK2135716130;received=172.23.39.108
From: <sip:2302@172.23.39.108:5060>;tag=790751827
To: "2303" <sip:2303@172.23.25.230>;tag=as53a52213
Call-ID: 0745ac96057c108a76dc3b61030c7c74@172.23.25.230:5060
CSeq: 8 BYE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:172.23.49.101:16398 --->


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

<--- SIP read from UDP:172.23.49.101:16398 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK415be79b
To: <sip:2301@172.23.49.101:16398;rinstance=0bec4a3d638cd5f0>;tag=4b21e62c
From: "2302"<sip:2302@172.23.25.230>;tag=as5da1659a
Call-ID: 40a9cc6d5cf722685ee961e10dd83ab3@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: eyeBeam release 1003l stamp 30936
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 480 "Temporarily Unavailable" back from 172.23.49.101:16398
Transmitting (no NAT) to 172.23.49.101:16398:
ACK sip:2301@172.23.49.101:16398;rinstance=0bec4a3d638cd5f0 SIP/2.0
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK415be79b
Max-Forwards: 70
From: "2302" <sip:2302@172.23.25.230>;tag=as5da1659a
To: <sip:2301@172.23.49.101:16398;rinstance=0bec4a3d638cd5f0>;tag=4b21e62c
Contact: <sip:2302@172.23.25.230:5060>
Call-ID: 40a9cc6d5cf722685ee961e10dd83ab3@172.23.25.230:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.0
Content-Length: 0


---
    -- SIP/2301-00000016 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [2301@milktest:2] Hangup("Local/2301@milktest-1e6d;2", "") in new stack
  == Spawn extension (milktest, 2301, 2) exited non-zero on 'Local/2301@milktest-1e6d;2'
[Nov 29 19:39:56] NOTICE[5621]: features.c:2114 builtin_atxfer: We're trying to call SIP/2302
  == Using SIP RTP CoS mark 5
Really destroying SIP dialog '40a9cc6d5cf722685ee961e10dd83ab3@172.23.25.230:5060' Method: INVITE
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 172.23.39.108:5060:
INVITE sip:2302@172.23.39.108:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK05d0eb77
Max-Forwards: 70
From: "2303" <sip:2303@172.23.25.230>;tag=as19e77a01
To: <sip:2302@172.23.39.108:5060>
Contact: <sip:2303@172.23.25.230:5060>
Call-ID: 2981c0d85be6e26d210e822d11585654@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.0
Date: Mon, 29 Nov 2010 11:39:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 1746162585 1746162585 IN IP4 172.23.25.230
s=Asterisk PBX 1.8.0
c=IN IP4 172.23.25.230
t=0 0
m=audio 13250 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Stopped music on hold on SIP/2303-00000014

<--- SIP read from UDP:172.23.39.108:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK05d0eb77
From: "2303" <sip:2303@172.23.25.230>;tag=as19e77a01
To: <sip:2302@172.23.39.108:5060>
Call-ID: 2981c0d85be6e26d210e822d11585654@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: ZyXEL V500-Series
Content-Length: 0

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

<--- SIP read from UDP:172.23.39.108:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK05d0eb77
From: "2303" <sip:2303@172.23.25.230>;tag=as19e77a01
To: <sip:2302@172.23.39.108:5060>;tag=557690693
Call-ID: 2981c0d85be6e26d210e822d11585654@172.23.25.230:5060
CSeq: 102 INVITE
User-Agent: ZyXEL V500-Series
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- SIP/2302-00000017 is ringing

<--- SIP read from UDP:172.23.39.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK05d0eb77
From: "2303" <sip:2303@172.23.25.230>;tag=as19e77a01
To: <sip:2302@172.23.39.108:5060>;tag=557690693
Call-ID: 2981c0d85be6e26d210e822d11585654@172.23.25.230:5060
CSeq: 102 INVITE
Contact: <sip:2302@172.23.39.108:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, PRACK, REFER, NOTIFY, INFO
User-Agent: ZyXEL V500-Series
Supported: timer100rel
Content-Length: 168

v=0
o=2302 2497 2497 IN IP4 172.23.39.108
s=Asterisk PBX 1.8.0
c=IN IP4 172.23.39.108
t=0 0
m=audio 50000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 172.23.39.108:50000
list_route: hop: <sip:2302@172.23.39.108:5060>
set_destination: Parsing <sip:2302@172.23.39.108:5060> for address/port to send to
set_destination: set destination to 172.23.39.108:5060
Transmitting (no NAT) to 172.23.39.108:5060:
ACK sip:2302@172.23.39.108:5060 SIP/2.0
Via: SIP/2.0/UDP 172.23.25.230:5060;branch=z9hG4bK5c9c02ef
Max-Forwards: 70
From: "2303" <sip:2303@172.23.25.230>;tag=as19e77a01
To: <sip:2302@172.23.39.108:5060>;tag=557690693
Contact: <sip:2303@172.23.25.230:5060>
Call-ID: 2981c0d85be6e26d210e822d11585654@172.23.25.230:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.0
Content-Length: 0


---
    -- <SIP/2302-00000017> Playing 'beep.ulaw' (language 'en')
    -- Locally bridging SIP/2303-00000014 and SIP/2302-00000017