rendered paste body<------------>
-- Native bridging SIP/10-081d0338 and SIP/20-081caad8
set_destination: Parsing <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c> for address/port to send to
set_destination: set destination to 10.0.76.69, port 20952
Audio is at 10.0.76.248 port 16076
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.76.69:20952:
INVITE sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c SIP/2.0
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK75d3ceb0;rport
From: "10" <sip:10@10.0.76.248>;tag=as2c89c807
To: <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c>;tag=6b72c313
Contact: <sip:10@10.0.76.248>
Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248
CSeq: 103 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: 232
v=0
o=root 3725 3726 IN IP4 10.0.0.62
s=session
c=IN IP4 10.0.0.62
t=0 0
m=audio 11248 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
astserv1*CLI>
<--- SIP read from 10.0.76.69:20952 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK75d3ceb0;rport=5060
Contact: <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c>
To: <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c>;tag=6b72c313
From: "10"<sip:10@10.0.76.248>;tag=as2c89c807
Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 178
v=0
o=- 0 2 IN IP4 10.0.76.69
s=CounterPath X-Lite 3.0
c=IN IP4 10.0.76.69
t=0 0
m=audio 6788 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 10.0.76.69:6788
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 10.0.76.69:6788
set_destination: Parsing <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c> for address/port to send to
set_destination: set destination to 10.0.76.69, port 20952
Transmitting (no NAT) to 10.0.76.69:20952:
ACK sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c SIP/2.0
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK604fdb65;rport
From: "10" <sip:10@10.0.76.248>;tag=as2c89c807
To: <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c>;tag=6b72c313
Contact: <sip:10@10.0.76.248>
Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
astserv1*CLI>
<--- SIP read from 10.0.0.62:42964 --->
ACK sip:20@10.0.76.248 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.62:42964;branch=z9hG4bK-d87543-f4771065a06ace7d-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:10@10.0.0.62:42964>
To: "20 (Softphone)"<sip:20@10.0.76.248>;tag=as5c96f43a
From: <sip:10@10.0.76.248>;tag=c3042d00
Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.
CSeq: 2 ACK
Proxy-Authorization: Digest username="10",realm="asterisk",nonce="3424cabd",uri="sip:20@10.0.76.248",response="934fece6dd50b62ce9d66b3ec73034a1",algorithm=MD5
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:10@10.0.0.62:42964> for address/port to send to
set_destination: set destination to 10.0.0.62, port 42964
Audio is at 10.0.76.248 port 11760
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.62:42964:
INVITE sip:10@10.0.0.62:42964 SIP/2.0
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK7c87e802;rport
From: "20 (Softphone)"<sip:20@10.0.76.248>;tag=as5c96f43a
To: <sip:10@10.0.76.248>;tag=c3042d00
Contact: <sip:20@10.0.76.248>
Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.
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: 197
v=0
o=root 3725 3727 IN IP4 10.0.76.69
s=session
c=IN IP4 10.0.76.69
t=0 0
m=audio 6788 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
astserv1*CLI>
<--- SIP read from 10.0.0.62:42964 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK7c87e802;rport=5060
To: <sip:10@10.0.76.248>;tag=c3042d00
From: "20 (Softphone)"<sip:20@10.0.76.248>;tag=as5c96f43a
Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Warning: 305 devnull "SDP: Incompatible media format: no common codec."
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:10@10.0.0.62:42964> for address/port to send to
set_destination: set destination to 10.0.0.62, port 42964
Transmitting (no NAT) to 10.0.0.62:42964:
ACK sip:10@10.0.0.62:42964 SIP/2.0
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK7c87e802;rport
From: "20 (Softphone)"<sip:20@10.0.76.248>;tag=as5c96f43a
To: <sip:10@10.0.76.248>;tag=c3042d00
Contact: <sip:20@10.0.76.248>
Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '2384eb1945bf6b1f77c07de73b35a260@10.0.76.248' in 6528 ms (Method: INVITE)
set_destination: Parsing <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c> for address/port to send to
set_destination: set destination to 10.0.76.69, port 20952
Reliably Transmitting (no NAT) to 10.0.76.69:20952:
BYE sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c SIP/2.0
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK3502836a;rport
From: "10" <sip:10@10.0.76.248>;tag=as2c89c807
To: <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c>;tag=6b72c313
Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (htl3r, 20, 1) exited non-zero on 'SIP/10-081d0338'
Scheduling destruction of SIP dialog 'YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:10@10.0.0.62:42964> for address/port to send to
set_destination: set destination to 10.0.0.62, port 42964
Reliably Transmitting (no NAT) to 10.0.0.62:42964:
BYE sip:10@10.0.0.62:42964 SIP/2.0
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK4b379144;rport
From: "20 (Softphone)"<sip:20@10.0.76.248>;tag=as5c96f43a
To: <sip:10@10.0.76.248>;tag=c3042d00
Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
astserv1*CLI>
<--- SIP read from 10.0.76.69:20952 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK3502836a;rport=5060
Contact: <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c>
To: <sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c>;tag=6b72c313
From: "10"<sip:10@10.0.76.248>;tag=as2c89c807
Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248
CSeq: 104 BYE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2384eb1945bf6b1f77c07de73b35a260@10.0.76.248' Method: INVITE
astserv1*CLI>
<--- SIP read from 10.0.0.62:42964 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK4b379144;rport=5060
Contact: <sip:10@10.0.0.62:42964>
To: <sip:10@10.0.76.248>;tag=c3042d00
From: "20 (Softphone)"<sip:20@10.0.76.248>;tag=as5c96f43a
Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.
CSeq: 103 BYE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.' Method: ACK