rendered paste body
---
-- Called 23003
voip*CLI>
<--- SIP read from 89.35.79.69:16088 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK201b6b8c;rport=5060
Contact: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=b47ae525
From: "ORAS"<sip:748881658@89.35.79.5>;tag=as5b05e2af
Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5
CSeq: 102 INVITE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/23003-08228400 is ringing
<--- Transmitting (no NAT) to 10.0.0.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15da6-555ee58-48bb;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-15da6-555ee58-14a
To: <sip:23003@10.0.0.2>;tag=as03e3b4ce
Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1
CSeq: 1 INVITE
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
<------------>
Audio is at 10.0.0.2 port 14776
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
<--- Transmitting (no NAT) to 10.0.0.1:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15da6-555ee58-48bb;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-15da6-555ee58-14a
To: <sip:23003@10.0.0.2>;tag=as03e3b4ce
Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 24756 24756 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 14776 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 89.35.79.69:16088 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK201b6b8c;rport=5060
Contact: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=b47ae525
From: "ORAS"<sip:748881658@89.35.79.5>;tag=as5b05e2af
Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5
CSeq: 102 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: 181
v=0
o=- 9 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 16992 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 89.35.79.69:16992
Found description format telephone-event for ID 101
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 89.35.79.69:16992
list_route: hop: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>
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=z9hG4bK47a7f252;rport
From: "ORAS" <sip:748881658@89.35.79.5>;tag=as5b05e2af
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=b47ae525
Contact: <sip:748881658@89.35.79.5>
Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- Call on SIP/23003-08228400 left from hold
-- SIP/23003-08228400 answered SIP/5060-08232388
Audio is at 10.0.0.2 port 14776
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 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15da6-555ee58-48bb;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-15da6-555ee58-14a
To: <sip:23003@10.0.0.2>;tag=as03e3b4ce
Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 24756 24757 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 14776 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
<------------>
-- Packet2Packet bridging SIP/5060-08232388 and SIP/23003-08228400
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=a000001-13c4-15da6-555ee58-14a
To: <sip:23003@10.0.0.2>;tag=as03e3b4ce
Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15dae-5560ed0-21d8
Max-Forwards: 70
Contact: <sip:748881658@10.0.0.1:5060;transport=UDP>
Content-Length:0
<------------->
--- (9 headers 0 lines) ---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
BYE sip:23003@10.0.0.2;TCID-27 SIP/2.0
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-15da6-555ee58-14a
To: <sip:23003@10.0.0.2>;tag=as03e3b4ce
Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1
CSeq: 2 BYE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-15dae-5560ed6-6816
Max-Forwards: 70
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
Sending to 10.0.0.1 : 5060 (no NAT)
<--- 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-15dae-5560ed6-6816;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-15da6-555ee58-14a
To: <sip:23003@10.0.0.2>;tag=as03e3b4ce
Call-ID: 97949c8c-a000001-13c4-15da6-555ee58-42ec@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 '1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5' in 32000 ms (Method: INVITE)
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=z9hG4bK2bb66b0b;rport
From: "ORAS" <sip:748881658@89.35.79.5>;tag=as5b05e2af
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=b47ae525
Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (default, 23003, 1) exited non-zero on 'SIP/5060-08232388'
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=z9hG4bK2bb66b0b;rport=5060
Contact: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=b47ae525
From: "ORAS"<sip:748881658@89.35.79.5>;tag=as5b05e2af
Call-ID: 1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5
CSeq: 103 BYE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1a0bbfa340b8ead330884abf6a4044fe@89.35.79.5' Method: INVITE
Really destroying SIP dialog '97949c8c-a000001-13c4-15da6-555ee58-42ec@10.0.0.1' Method: BYE
Really destroying SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' Method: REGISTER
voip*CLI>
<--- SIP read from 89.35.79.69:16088 --->
<------------->
--- (0 headers 1 lines) ---
voip*CLI>