<--- SIP read from 84.8.191.13:5060 --->
INVITE sip:*7702070325205@84.8.129.188;user=phone SIP/2.0
Max-Forwards: 138
Session-Expires: 1800;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:5205@84.8.129.188:5060;user=phone>
From: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
P-Asserted-Identity: <sip:7973466309@10.40.126.198;user=phone>
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bK645fbc1b08cd36674c549db2e33e8d5a
Contact: <sip:07973466309@84.8.191.13:5060;user=phone>
Call-Info: <sip:84.8.191.13>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 347

v=0
o=aosbc1 2147483647 2147483647 IN IP4 84.8.191.13
s=sip call
c=IN IP4 10.40.126.198
t=0 0
m=audio 40420 RTP/AVP 8 18
a=ptime:10
a=fmtp:18 annexb=no
m=image 40422 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (16 headers 15 lines) ---
Sending to 84.8.191.13 : 5060 (no NAT)
Using INVITE request as basis request - 644-3462881968-720466@aosbc1.alwaysongroup.com
Found peer 'nextpoint-sbc'
Found RTP audio format 8
Found RTP audio format 18
[Sep 25 16:27:15] WARNING[13765]: chan_sip.c:5159 process_sdp: Unsupported SDP media type in offer: image 40422 udptl t38
Peer audio RTP is at port 10.40.126.198:40420
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=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 10.40.126.198:40420
Looking for *7702070325205 in default (domain 84.8.129.188)
list_route: hop: <sip:07973466309@84.8.191.13:5060;user=phone>
aovastest01*CLI>
<--- Transmitting (no NAT) to 84.8.191.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bK645fbc1b08cd36674c549db2e33e8d5a;received=84.8.191.13
From: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
To: <sip:5205@84.8.129.188:5060;user=phone>
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 1 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*7702070325205@84.8.129.188>
Content-Length: 0


<------------>
Audio is at 84.8.129.188 port 13156
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 195.219.133.219:5060:
INVITE sip:02070325205@195.219.133.219 SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK072cc1a5;rport
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>
Contact: <sip:07973466309@84.8.129.188>
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 102 INVITE
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07973466309" <sip:07973466309@84.8.129.188>;privacy=off;screen=no
Date: Fri, 25 Sep 2009 15:27:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 13733 13733 IN IP4 84.8.129.188
s=session
c=IN IP4 84.8.129.188
t=0 0
m=audio 13156 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK072cc1a5;rport
CSeq: 102 INVITE
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 195.219.133.219:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK072cc1a5;rport
CSeq: 102 INVITE
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:02070325205@195.219.133.219:5065;transport=udp>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- Transmitting (no NAT) to 84.8.191.13:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bK645fbc1b08cd36674c549db2e33e8d5a;received=84.8.191.13
From: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
To: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 1 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*7702070325205@84.8.129.188>
Content-Length: 0


<------------>
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK072cc1a5;rport
CSeq: 102 INVITE
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:02070325205@195.219.133.219:5065;transport=udp>
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY
Content-Length: 197

v=0
o=- 4057692265 4057692265 IN IP4 195.219.133.219
s=-
c=IN IP4 84.8.129.165
t=0 0
a=sendrecv
m=audio 5018 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtcp:5019

<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Peer audio RTP is at port 84.8.129.165:5018
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
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 - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 84.8.129.165:5018
list_route: hop: <sip:02070325205@195.219.133.219:5065;transport=udp>
set_destination: Parsing <sip:02070325205@195.219.133.219:5065;transport=udp> for address/port to send to
set_destination: set destination to 195.219.133.219, port 5065
Transmitting (no NAT) to 195.219.133.219:5065:
ACK sip:02070325205@195.219.133.219:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK4e18a79c;rport
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:07973466309@84.8.129.188>
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 102 ACK
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07973466309" <sip:07973466309@84.8.129.188>;privacy=off;screen=no
Content-Length: 0


---
Audio is at 84.8.129.188 port 10118
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 84.8.191.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bK645fbc1b08cd36674c549db2e33e8d5a;received=84.8.191.13
From: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
To: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 1 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*7702070325205@84.8.129.188>
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 13733 13733 IN IP4 84.8.129.188
s=session
c=IN IP4 84.8.129.188
t=0 0
m=audio 10118 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:10
a=sendrecv

<------------>
set_destination: Parsing <sip:02070325205@195.219.133.219:5065;transport=udp> for address/port to send to
set_destination: set destination to 195.219.133.219, port 5065
Audio is at 84.8.129.188 port 13156
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 195.219.133.219:5065:
INVITE sip:02070325205@195.219.133.219:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK3bc15a98;rport
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:07973466309@84.8.129.188>
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 103 INVITE
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07973466309" <sip:07973466309@84.8.129.188>;privacy=off;screen=no
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: 186

v=0
o=root 13733 13734 IN IP4 10.40.126.198
s=session
c=IN IP4 10.40.126.198
t=0 0
m=audio 40420 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK3bc15a98;rport
CSeq: 103 INVITE
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 84.8.191.13:5060 --->
ACK sip:*7702070325205@84.8.129.188 SIP/2.0
Max-Forwards: 68
To: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
From: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 1 ACK
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bKf8d3bc210a7fd22775ba444e20f4fbb7
Contact: <sip:07973466309@84.8.191.13:5060;user=phone>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:07973466309@84.8.191.13:5060;user=phone> for address/port to send to
set_destination: set destination to 84.8.191.13, port 5060
Audio is at 84.8.129.188 port 10118
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 84.8.191.13:5060:
INVITE sip:07973466309@84.8.191.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK4dc8af69;rport
From: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
To: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
Contact: <sip:*7702070325205@84.8.129.188>
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 102 INVITE
User-Agent: alwaysON
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: 183

v=0
o=root 13733 13734 IN IP4 84.8.129.165
s=session
c=IN IP4 84.8.129.165
t=0 0
m=audio 5018 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:10
a=sendrecv

---
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK3bc15a98;rport
CSeq: 103 INVITE
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:02070325205@195.219.133.219:5065;transport=udp>
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY
Content-Length: 173

v=0
o=- 4057692265 4057692266 IN IP4 195.219.133.219
s=-
c=IN IP4 84.8.129.165
t=0 0
a=sendrecv
m=audio 5018 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtcp:5019

<------------->
--- (10 headers 10 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 84.8.129.165:5018
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=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 84.8.129.165:5018
set_destination: Parsing <sip:02070325205@195.219.133.219:5065;transport=udp> for address/port to send to
set_destination: set destination to 195.219.133.219, port 5065
Transmitting (no NAT) to 195.219.133.219:5065:
ACK sip:02070325205@195.219.133.219:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK1a74707a;rport
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:07973466309@84.8.129.188>
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 103 ACK
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07973466309" <sip:07973466309@84.8.129.188>;privacy=off;screen=no
Content-Length: 0


---
aovastest01*CLI>
<--- SIP read from 84.8.191.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK4dc8af69;rport
From: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
To: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 84.8.191.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK4dc8af69;rport
To: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
From: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 102 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:07973466309@84.8.191.13:5060;user=phone>
Call-Info: <sip:84.8.191.13>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 347

v=0
o=aosbc1 2147483647 2147483647 IN IP4 84.8.191.13
s=sip call
c=IN IP4 10.40.126.198
t=0 0
m=audio 40420 RTP/AVP 8 18
a=ptime:10
a=fmtp:18 annexb=no
m=image 40422 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (11 headers 15 lines) ---
Found RTP audio format 8
Found RTP audio format 18
[Sep 25 16:27:22] WARNING[13765]: chan_sip.c:5159 process_sdp: Unsupported SDP media type in offer: image 40422 udptl t38
Peer audio RTP is at port 10.40.126.198:40420
Capabilities: us - 0x8 (alaw), peer - audio=0x108 (alaw|g729)/video=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 10.40.126.198:40420
set_destination: Parsing <sip:07973466309@84.8.191.13:5060;user=phone> for address/port to send to
set_destination: set destination to 84.8.191.13, port 5060
Transmitting (no NAT) to 84.8.191.13:5060:
ACK sip:07973466309@84.8.191.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK47976fb3;rport
From: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
To: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
Contact: <sip:*7702070325205@84.8.129.188>
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 102 ACK
User-Agent: alwaysON
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:07973466309@84.8.191.13:5060;user=phone> for address/port to send to
set_destination: set destination to 84.8.191.13, port 5060
Audio is at 84.8.129.188 port 10118
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 84.8.191.13:5060:
INVITE sip:07973466309@84.8.191.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK496bf01b;rport
From: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
To: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
Contact: <sip:*7702070325205@84.8.129.188>
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 103 INVITE
User-Agent: alwaysON
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: 183

v=0
o=root 13733 13735 IN IP4 84.8.129.165
s=session
c=IN IP4 84.8.129.165
t=0 0
m=audio 5018 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:10
a=sendrecv

---
aovastest01*CLI>
<--- SIP read from 84.8.191.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK496bf01b;rport
From: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
To: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 103 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 84.8.191.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK496bf01b;rport
To: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
From: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 103 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:07973466309@84.8.191.13:5060;user=phone>
Call-Info: <sip:84.8.191.13>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 347

v=0
o=aosbc1 2147483647 2147483647 IN IP4 84.8.191.13
s=sip call
c=IN IP4 10.40.126.198
t=0 0
m=audio 40420 RTP/AVP 8 18
a=ptime:10
a=fmtp:18 annexb=no
m=image 40422 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (11 headers 15 lines) ---
Found RTP audio format 8
Found RTP audio format 18
[Sep 25 16:27:22] WARNING[13765]: chan_sip.c:5159 process_sdp: Unsupported SDP media type in offer: image 40422 udptl t38
Peer audio RTP is at port 10.40.126.198:40420
Capabilities: us - 0x8 (alaw), peer - audio=0x108 (alaw|g729)/video=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 10.40.126.198:40420
set_destination: Parsing <sip:07973466309@84.8.191.13:5060;user=phone> for address/port to send to
set_destination: set destination to 84.8.191.13, port 5060
Transmitting (no NAT) to 84.8.191.13:5060:
ACK sip:07973466309@84.8.191.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK28a70542;rport
From: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
To: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
Contact: <sip:*7702070325205@84.8.129.188>
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 103 ACK
User-Agent: alwaysON
Max-Forwards: 70
Content-Length: 0


---
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
INVITE sip:07973466309@84.8.129.188 SIP/2.0
From: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
To: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 1 INVITE
Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK194494549752754
Contact: <sip:02070325205@195.219.133.219:5065;transport=udp>
Max-Forwards: 70
Content-Type: application/sdp
Remote-Party-ID: "Test,Easynet" <sip:02070325205@195.219.133.219>;party=calling;id-type=subscriber;privacy=off
P-Asserted-Identity: "Test,Easynet" <sip:alwaysonvpn.com>
Content-Length: 144

v=0
o=- 4057692265 4057692267 IN IP4 195.219.133.219
s=SIP Call
c=IN IP4 195.219.151.7
t=0 0
m=audio 1108 RTP/AVP 8
a=rtpmap:8 pcma/8000

<------------->
--- (12 headers 7 lines) ---
Sending to 195.219.133.219 : 5065 (no NAT)
Found RTP audio format 8
Peer audio RTP is at port 195.219.151.7:1108
Found audio description format pcma for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=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 195.219.151.7:1108

<--- Transmitting (no NAT) to 195.219.133.219:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK194494549752754;received=195.219.133.219
From: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
To: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 1 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07973466309@84.8.129.188>
Content-Length: 0


<------------>
Audio is at 84.8.129.188 port 13156
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 195.219.133.219:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK194494549752754;received=195.219.133.219
From: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
To: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 1 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07973466309@84.8.129.188>
Content-Type: application/sdp
Content-Length: 186

v=0
o=root 13733 13735 IN IP4 10.40.126.198
s=session
c=IN IP4 10.40.126.198
t=0 0
m=audio 40420 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
ACK sip:07973466309@84.8.129.188 SIP/2.0
From: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
To: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 1 ACK
Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK1008356288775929
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from 84.8.191.13:5060 --->
OPTIONS sip:84.8.129.188:5060 SIP/2.0
Max-Forwards: 70
To: <sip:84.8.129.188>
From: <sip:84.8.129.188>;tag=3462881982-258160
Call-ID: 656-3462881982-258159@aosbc1.alwaysongroup.com
CSeq: 1 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bKae413f092ce525d13f8cd099de12cad0
Contact: <sip:84.8.191.13:5060>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Looking for s in default (domain 84.8.129.188)
aovastest01*CLI>
<--- Transmitting (no NAT) to 84.8.191.13:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bKae413f092ce525d13f8cd099de12cad0;received=84.8.191.13
From: <sip:84.8.129.188>;tag=3462881982-258160
To: <sip:84.8.129.188>;tag=as2930ce63
Call-ID: 656-3462881982-258159@aosbc1.alwaysongroup.com
CSeq: 1 OPTIONS
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '656-3462881982-258159@aosbc1.alwaysongroup.com' in 32000 ms (Method: OPTIONS)
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
INVITE sip:07973466309@84.8.129.188 SIP/2.0
From: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
To: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 2 INVITE
Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK704491186928261
Contact: <sip:02070325205@195.219.133.219:5065;transport=udp>
Max-Forwards: 70
Content-Type: application/sdp
Remote-Party-ID: "NA" <sip:02070325205@195.219.133.219>;party=calling;id-type=subscriber;privacy=off
P-Asserted-Identity: sip:02070325205@195.219.133.219
Content-Length: 163

v=0
o=- 4057692265 4057692268 IN IP4 195.219.133.219
s=Polycom IP Phone
c=IN IP4 84.8.129.140
t=0 0
m=audio 2242 RTP/AVP 8
a=sendrecv
a=rtpmap:8 PCMA/8000

<------------->
--- (12 headers 8 lines) ---
Sending to 195.219.133.219 : 5065 (no NAT)
Found RTP audio format 8
Peer audio RTP is at port 84.8.129.140:2242
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=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 84.8.129.140:2242
aovastest01*CLI>
<--- Transmitting (no NAT) to 195.219.133.219:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK704491186928261;received=195.219.133.219
From: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
To: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 2 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07973466309@84.8.129.188>
Content-Length: 0


<------------>
Audio is at 84.8.129.188 port 13156
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 195.219.133.219:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK704491186928261;received=195.219.133.219
From: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
To: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 2 INVITE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:07973466309@84.8.129.188>
Content-Type: application/sdp
Content-Length: 186

v=0
o=root 13733 13736 IN IP4 10.40.126.198
s=session
c=IN IP4 10.40.126.198
t=0 0
m=audio 40420 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
ACK sip:07973466309@84.8.129.188 SIP/2.0
From: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
To: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 2 ACK
Via: SIP/2.0/UDP 195.219.133.219:5065;branch=z9hG4bK1555547337951400
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from 84.8.191.13:5060 --->
BYE sip:*7702070325205@84.8.129.188 SIP/2.0
Max-Forwards: 68
To: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
From: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 2 BYE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bK24d3e5f94f774166ba6aa4a20baa634f
Contact: <sip:07973466309@84.8.191.13:5060;user=phone>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 84.8.191.13 : 5060 (no NAT)

<--- Transmitting (no NAT) to 84.8.191.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.8.191.13:5060;branch=z9hG4bK24d3e5f94f774166ba6aa4a20baa634f;received=84.8.191.13
From: <sip:07973466309@84.8.191.13>;tag=3462881968-720471
To: <sip:5205@84.8.129.188:5060;user=phone>;tag=as02ef75aa
Call-ID: 644-3462881968-720466@aosbc1.alwaysongroup.com
CSeq: 2 BYE
User-Agent: alwaysON
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*7702070325205@84.8.129.188>
Content-Length: 0


<------------>
set_destination: Parsing <sip:02070325205@195.219.133.219:5065;transport=udp> for address/port to send to
set_destination: set destination to 195.219.133.219, port 5065
Audio is at 84.8.129.188 port 13156
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 195.219.133.219:5065:
INVITE sip:02070325205@195.219.133.219:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK2e9c6840;rport
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:07973466309@84.8.129.188>
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 104 INVITE
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07973466309" <sip:07973466309@84.8.129.188>;privacy=off;screen=no
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: 184

v=0
o=root 13733 13737 IN IP4 84.8.129.188
s=session
c=IN IP4 84.8.129.188
t=0 0
m=audio 13156 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Scheduling destruction of SIP dialog '708511da46a033fb56d30f3643bfab96@84.8.129.188' in 32000 ms (Method: ACK)
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK2e9c6840;rport
CSeq: 104 INVITE
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '644-3462881968-720466@aosbc1.alwaysongroup.com' Method: BYE
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK2e9c6840;rport
CSeq: 104 INVITE
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:02070325205@195.219.133.219:5065;transport=udp>
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY
Content-Length: 163

v=0
o=- 4057692265 4057692268 IN IP4 195.219.133.219
s=Polycom IP Phone
c=IN IP4 84.8.129.140
t=0 0
m=audio 2242 RTP/AVP 8
a=sendrecv
a=rtpmap:8 PCMA/8000

<------------->
--- (10 headers 8 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 84.8.129.140:2242
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=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 84.8.129.140:2242
set_destination: Parsing <sip:02070325205@195.219.133.219:5065;transport=udp> for address/port to send to
set_destination: set destination to 195.219.133.219, port 5065
Transmitting (no NAT) to 195.219.133.219:5065:
ACK sip:02070325205@195.219.133.219:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK60079081;rport
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Contact: <sip:07973466309@84.8.129.188>
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 104 ACK
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07973466309" <sip:07973466309@84.8.129.188>;privacy=off;screen=no
Content-Length: 0


---
set_destination: Parsing <sip:02070325205@195.219.133.219:5065;transport=udp> for address/port to send to
set_destination: set destination to 195.219.133.219, port 5065
Reliably Transmitting (no NAT) to 195.219.133.219:5065:
BYE sip:02070325205@195.219.133.219:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK0f21fd6f;rport
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
CSeq: 105 BYE
User-Agent: alwaysON
Max-Forwards: 70
Remote-Party-ID: "07973466309" <sip:07973466309@84.8.129.188>;privacy=off;screen=no
Content-Length: 0


---
Scheduling destruction of SIP dialog '708511da46a033fb56d30f3643bfab96@84.8.129.188' in 32000 ms (Method: ACK)
aovastest01*CLI>
<--- SIP read from 195.219.133.219:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.8.129.188:5060;branch=z9hG4bK0f21fd6f;rport
CSeq: 105 BYE
Call-ID: 708511da46a033fb56d30f3643bfab96@84.8.129.188
From: "07973466309" <sip:07973466309@84.8.129.188>;tag=as6f644d28
To: <sip:02070325205@195.219.133.219>;tag=9c991fc8-1dd1-11b2-b849-b03162323164+9c991fc8
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '708511da46a033fb56d30f3643bfab96@84.8.129.188' Method: ACK
Really destroying SIP dialog '656-3462881982-258159@aosbc1.alwaysongroup.com' Method: OPTIONS