<--- SIP read from 172.16.16.132:5060 --->
INVITE sip:9000@voip.dogmatix.dnv.org;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bKe82b8935EBBD038C
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>
CSeq: 1 INVITE
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
Contact: <sip:2348-Polycom@172.16.16.132:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.6.0036
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 239

v=0
o=- 1164221670 1164221670 IN IP4 172.16.16.132
s=Polycom IP Phone
c=IN IP4 172.16.16.132
t=0 0
m=audio 2254 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (14 headers 10 lines) ---
Sending to 172.16.16.132 : 5060 (no NAT)
Using INVITE request as basis request - 6ed058b-e37d4981-980b7af2@172.16.16.132
Found peer '2348-Polycom'

<--- Reliably Transmitting (no NAT) to 172.16.16.132:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bKe82b8935EBBD038C;received=172.16.16.132
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>;tag=as75f25781
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
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="768e56a1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '6ed058b-e37d4981-980b7af2@172.16.16.132' in 5696 ms (Method: INVITE)
dogmatix*CLI> 
<--- SIP read from 172.16.16.132:5060 --->
ACK sip:9000@voip.dogmatix.dnv.org SIP/2.0
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bKe82b8935EBBD038C
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>;tag=as75f25781
CSeq: 1 ACK
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
Contact: <sip:2348-Polycom@172.16.16.132:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.6.0036
Max-Forwards: 70
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
dogmatix*CLI> 
<--- SIP read from 172.16.16.132:5060 --->
INVITE sip:9000@voip.dogmatix.dnv.org;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bKc5e51b9e1344A04D
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>
CSeq: 2 INVITE
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
Contact: <sip:2348-Polycom@172.16.16.132:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.6.0036
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="2348-Polycom", realm="asterisk", nonce="768e56a1", uri="sip:9000@voip.dogmatix.dnv.org;user=phone", response="0bd42b91048a78e46df25cd2342566ea", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 239

v=0
o=- 1164221670 1164221670 IN IP4 172.16.16.132
s=Polycom IP Phone
c=IN IP4 172.16.16.132
t=0 0
m=audio 2254 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (15 headers 10 lines) ---
Sending to 172.16.16.132 : 5060 (no NAT)
Using INVITE request as basis request - 6ed058b-e37d4981-980b7af2@172.16.16.132
Found peer '2348-Polycom'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.16.16.132:2254
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/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 172.16.16.132:2254
Looking for 9000 in long-distance (domain voip.dogmatix.dnv.org)
list_route: hop: <sip:2348-Polycom@172.16.16.132:5060>

<--- Transmitting (no NAT) to 172.16.16.132:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bKc5e51b9e1344A04D;received=172.16.16.132
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9000@172.16.16.12>
Content-Length: 0


<------------>
Audio is at 172.16.16.12 port 17906
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
dogmatix*CLI> 
<--- Reliably Transmitting (no NAT) to 172.16.16.132:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bKc5e51b9e1344A04D;received=172.16.16.132
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>;tag=as77fe0712
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9000@172.16.16.12>
Content-Type: application/sdp
Content-Length: 228

v=0
o=root 19541 19541 IN IP4 172.16.16.12
s=session
c=IN IP4 172.16.16.12
t=0 0
m=audio 17906 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

<------------>
dogmatix*CLI> 
<--- SIP read from 172.16.16.132:5060 --->
ACK sip:9000@172.16.16.12 SIP/2.0
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bK261abd993614C350
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>;tag=as77fe0712
CSeq: 2 ACK
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
Contact: <sip:2348-Polycom@172.16.16.132:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.6.0036
Proxy-Authorization: Digest username="2348-Polycom", realm="asterisk", nonce="768e56a1", uri="sip:9000@voip.dogmatix.dnv.org;user=phone", response="0bd42b91048a78e46df25cd2342566ea", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
dogmatix*CLI> 
<--- SIP read from 172.16.16.132:5060 --->
BYE sip:9000@172.16.16.12 SIP/2.0
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bK73da91af2E56AC76
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>;tag=as77fe0712
CSeq: 3 BYE
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
Contact: <sip:2348-Polycom@172.16.16.132:5060>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.6.0036
Proxy-Authorization: Digest username="2348-Polycom", realm="asterisk", nonce="768e56a1", uri="sip:9000@voip.dogmatix.dnv.org;user=phone", response="bb60f3269a93270996c9c5b83635f7e8", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 172.16.16.132 : 5060 (no NAT)

<--- Transmitting (no NAT) to 172.16.16.132:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.16.132:5060;branch=z9hG4bK73da91af2E56AC76;received=172.16.16.132
From: "2348-Polycom" <sip:2348-Polycom@voip.dogmatix.dnv.org>;tag=33448B17-DBC8C0F8
To: <sip:9000@voip.dogmatix.dnv.org;user=phone>;tag=as77fe0712
Call-ID: 6ed058b-e37d4981-980b7af2@172.16.16.132
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9000@172.16.16.12>
Content-Length: 0


<------------>
Really destroying SIP dialog '6ed058b-e37d4981-980b7af2@172.16.16.132' Method: BYE
Reliably Transmitting (no NAT) to 172.16.16.132:5060:
OPTIONS sip:2348-Polycom@172.16.16.132:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1c20214a;rport
From: "asterisk" <sip:asterisk@172.16.16.12>;tag=as4f60cfe3
To: <sip:2348-Polycom@172.16.16.132:5060>
Contact: <sip:asterisk@172.16.16.12>
Call-ID: 4afc68716edc675927b4528e2a63bd59@172.16.16.12
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 22 Nov 2006 18:55:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
dogmatix*CLI> 
<--- SIP read from 172.16.16.132:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1c20214a;rport
From: "asterisk" <sip:asterisk@172.16.16.12>;tag=as4f60cfe3
To: <sip:2348-Polycom@172.16.16.132:5060>;tag=60B40FA8-24DA0BB1
CSeq: 102 OPTIONS
Call-ID: 4afc68716edc675927b4528e2a63bd59@172.16.16.12
Contact: <sip:2348-Polycom@172.16.16.132:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.6.0036
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4afc68716edc675927b4528e2a63bd59@172.16.16.12' Method: OPTIONS