rendered paste bodya=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
ACK sip:23003@10.0.0.2 SIP/2.0
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-1632d-56b8808-6fa2
To: <sip:23003@10.0.0.2>;tag=as7085b194
Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16335-56ba6ea-53f0
Max-Forwards: 70
Contact: <sip:748881658@10.0.0.1:5060;transport=UDP>
Content-Length:0
<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:748881658@10.0.0.1:5060;transport=UDP> for address/port to send to
set_destination: set destination to 10.0.0.1, port 5060
Audio is at 10.0.0.2 port 11728
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.1:5060:
INVITE sip:748881658@10.0.0.1:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK195f8746;rport
From: <sip:23003@10.0.0.2>;tag=as7085b194
To: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-1632d-56b8808-6fa2
Contact: <sip:23003@10.0.0.2>
Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1
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: 306
v=0
o=root 24756 24757 IN IP4 89.35.79.69
s=session
c=IN IP4 89.35.79.69
t=0 0
m=audio 18638 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
BYE sip:23003@10.0.0.2;TCID-28 SIP/2.0
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-1632d-56b8808-6fa2
To: <sip:23003@10.0.0.2>;tag=as7085b194
Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1
CSeq: 2 BYE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16335-56ba6f2-63ee
Max-Forwards: 70
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
Sending to 10.0.0.1 : 5060 (no NAT)
voip*CLI>
<--- Transmitting (no NAT) to 10.0.0.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16335-56ba6f2-63ee;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-1632d-56b8808-6fa2
To: <sip:23003@10.0.0.2>;tag=as7085b194
Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '5cef322324cadbc0372e39e17409242f@89.35.79.5' in 32000 ms (Method: INVITE)
== Spawn extension (default, 23003, 1) exited non-zero on 'SIP/5060-0822ee68'
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 400 Bad Request
From: <sip:23003@10.0.0.2>;tag=as7085b194
To: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-1632d-56b8808-6fa2
Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK195f8746;rport
Content-Length:0
<------------->
--- (7 headers 0 lines) ---
-- Got SIP response 400 "Bad Request" back from 10.0.0.1
set_destination: Parsing <sip:748881658@10.0.0.1:5060;transport=UDP> for address/port to send to
set_destination: set destination to 10.0.0.1, port 5060
Transmitting (no NAT) to 10.0.0.1:5060:
ACK sip:748881658@10.0.0.1:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK195f8746;rport
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-1632d-56b8808-6fa2
To: <sip:23003@10.0.0.2>;tag=as7085b194
Contact: <sip:23003@10.0.0.2>
Call-ID: 97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Really destroying SIP dialog '97949dbc-a000001-13c4-1632d-56b8808-384@10.0.0.1' Method: BYE
voip*CLI>
<--- SIP read from 89.35.79.69:16088 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK0714407e;rport=5060
Contact: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=8d028934
From: "ORAS"<sip:748881658@89.35.79.5>;tag=as79772655
Call-ID: 5cef322324cadbc0372e39e17409242f@89.35.79.5
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
ser-Agent: X-Lite release 1006e stamp 34025
Content-Length: 178
v=0
o=- 7 2 IN IP4 89.35.79.69
s=CounterPath X-Lite 3.0
c=IN IP4 89.35.79.69
t=0 0
m=audio 18638 RTP/AVP 8 97
a=fmtp:97 0-15
a=rtpmap:97 telephone-event/8000
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 97
Peer audio RTP is at port 89.35.79.69:18638
Found description format telephone-event for ID 97
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 89.35.79.69:18638
set_destination: Parsing <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab> for address/port to send to
set_destination: set destination to 89.35.79.69, port 16088
Transmitting (no NAT) to 89.35.79.69:16088:
ACK sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK717e4fe7;rport
From: "ORAS" <sip:748881658@89.35.79.5>;tag=as79772655
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=8d028934
Contact: <sip:748881658@89.35.79.5>
Call-ID: 5cef322324cadbc0372e39e17409242f@89.35.79.5
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
set_destination: Parsing <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab> for address/port to send to
set_destination: set destination to 89.35.79.69, port 16088
Reliably Transmitting (no NAT) to 89.35.79.69:16088:
BYE sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK3dd48e88;rport
From: "ORAS" <sip:748881658@89.35.79.5>;tag=as79772655
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=8d028934
Call-ID: 5cef322324cadbc0372e39e17409242f@89.35.79.5
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '5cef322324cadbc0372e39e17409242f@89.35.79.5' in 32000 ms (Method: INVITE)
voip*CLI>
<--- SIP read from 89.35.79.69:16088 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK3dd48e88;rport=5060
Contact: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=8d028934
From: "ORAS"<sip:748881658@89.35.79.5>;tag=as79772655
Call-ID: 5cef322324cadbc0372e39e17409242f@89.35.79.5
CSeq: 104 BYE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '5cef322324cadbc0372e39e17409242f@89.35.79.5' Method: INVITE
voip*CLI>
<--- SIP read from 89.35.79.69:16088 --->
<------------->
--- (0 headers 1 lines) ---
Scheduling destruction of SIP dialog '323002043e33719a3a7fed292556809b@89.35.79.5' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 89.35.79.247:5060:
NOTIFY sip:23001@89.35.79.247:5060 SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK3ea55859;rport
From: "asterisk" <sip:asterisk@89.35.79.5>;tag=as78ea5c15
To: <sip:23001@89.35.79.247:5060>
Contact: <sip:asterisk@89.35.79.5>
Call-ID: 323002043e33719a3a7fed292556809b@89.35.79.5
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 91
Messages-Waiting: yes
Message-Account: sip:asterisk@89.35.79.5
Voice-Message: 1/0 (0/0)
---
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK3ea55859;rport
Call-ID: 323002043e33719a3a7fed292556809b@89.35.79.5
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk@89.35.79.5>;tag=as78ea5c15
To: <sip:23001@89.35.79.247:5060>;tag=uq6IKSs1wDgqoZpt
Contact: <sip:23001@89.35.79.247:5060>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '323002043e33719a3a7fed292556809b@89.35.79.5' Method: NOTIFY
voip*CLI>