rendered paste bodyakira*CLI>
<--- SIP read from 10.0.1.41:5060 --->
INVITE sip:901489574204@10.0.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-194e1a59
From: <sip:3101@10.0.1.21>;tag=b2137b34817f28b9o0
To: <sip:901489574204@10.0.1.21>
Call-ID: 30c12d10-963bc08d@10.0.1.41
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="3352271d",uri="sip:901489574204@10.0.1.21",algorithm=MD5,response="39814b89d35d1938a3900fb67b17c257"
Contact: <sip:3101@10.0.1.41:5060>
Expires: 240
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 391
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 1333214 1333214 IN IP4 10.0.1.41
s=-
c=IN IP4 10.0.1.41
t=0 0
m=audio 16394 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.0.1.41 : 5060 (no NAT)
Using INVITE request as basis request - 30c12d10-963bc08d@10.0.1.41
Found user '3101'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 10.0.1.41:16394
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format G729a for ID 18
Found description format G726-40 for ID 96
Found description format G726-24 for ID 97
Found description format G726-16 for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/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 10.0.1.41:16394
Looking for 901489574204 in handsets (domain 10.0.1.21)
list_route: hop: <sip:3101@10.0.1.41:5060>
<--- Transmitting (no NAT) to 10.0.1.41:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-194e1a59;received=10.0.1.41
From: <sip:3101@10.0.1.21>;tag=b2137b34817f28b9o0
To: <sip:901489574204@10.0.1.21>
Call-ID: 30c12d10-963bc08d@10.0.1.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:901489574204@10.0.1.21>
Content-Length: 0
<------------>
-- Executing [901489574204@handsets:1] Dial("SIP/3101-081e1220", "SIP/sipuraline1/901489574204|60") in new stack
Audio is at 10.0.1.21 port 18352
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.40:5061:
INVITE sip:901489574204@10.0.1.40:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK671dcf5c;rport
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as7f28049e
To: <sip:901489574204@10.0.1.40:5061>
Contact: <sip:3101@10.0.1.21>
Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 21 Dec 2007 13:37:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 23787 23787 IN IP4 10.0.1.21
s=session
c=IN IP4 10.0.1.21
t=0 0
m=audio 18352 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
---
-- Called sipuraline1/901489574204
akira*CLI>
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 100 Trying
To: <sip:901489574204@10.0.1.40:5061>
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as7f28049e
Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK671dcf5c
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
akira*CLI>
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 180 Ringing
To: <sip:901489574204@10.0.1.40:5061>;tag=52de3dd8aa3e8526i1
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as7f28049e
Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK671dcf5c
Server: Linksys/SPA3102-3.3.6(GW)
Remote-Party-ID: sipuraline1 <sip:sipuraline1@10.0.1.21>;screen=yes;party=called
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/sipuraline1-081e5188 is ringing
<--- Transmitting (no NAT) to 10.0.1.41:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-194e1a59;received=10.0.1.41
From: <sip:3101@10.0.1.21>;tag=b2137b34817f28b9o0
To: <sip:901489574204@10.0.1.21>;tag=as2fd54815
Call-ID: 30c12d10-963bc08d@10.0.1.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:901489574204@10.0.1.21>
Content-Length: 0
<------------>
akira*CLI>
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 200 OK
To: <sip:901489574204@10.0.1.40:5061>;tag=52de3dd8aa3e8526i1
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as7f28049e
Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK671dcf5c
Contact: sipuraline1 <sip:901489574204@10.0.1.40:5061>
Server: Linksys/SPA3102-3.3.6(GW)
Remote-Party-ID: sipuraline1 <sip:sipuraline1@10.0.1.21>;screen=yes;party=called
Content-Length: 247
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 145055 145055 IN IP4 10.0.1.40
s=-
c=IN IP4 10.0.1.40
t=0 0
m=audio 16422 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 10.0.1.40:16422
Found description format PCMA for ID 8
Found description format NSE for ID 100
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 10.0.1.40:16422
list_route: hop: <sip:901489574204@10.0.1.40:5061>
set_destination: Parsing <sip:901489574204@10.0.1.40:5061> for address/port to send to
set_destination: set destination to 10.0.1.40, port 5061
Transmitting (no NAT) to 10.0.1.40:5061:
ACK sip:901489574204@10.0.1.40:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK411b2875;rport
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as7f28049e
To: <sip:901489574204@10.0.1.40:5061>;tag=52de3dd8aa3e8526i1
Contact: <sip:3101@10.0.1.21>
Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/sipuraline1-081e5188 answered SIP/3101-081e1220
Audio is at 10.0.1.21 port 14836
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.1.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-194e1a59;received=10.0.1.41
From: <sip:3101@10.0.1.21>;tag=b2137b34817f28b9o0
To: <sip:901489574204@10.0.1.21>;tag=as2fd54815
Call-ID: 30c12d10-963bc08d@10.0.1.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:901489574204@10.0.1.21>
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 23787 23787 IN IP4 10.0.1.21
s=session
c=IN IP4 10.0.1.21
t=0 0
m=audio 14836 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
<------------>
-- Packet2Packet bridging SIP/3101-081e1220 and SIP/sipuraline1-081e5188
akira*CLI>
<--- SIP read from 10.0.1.41:5060 --->
ACK sip:901489574204@10.0.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-4eba0be2
From: <sip:3101@10.0.1.21>;tag=b2137b34817f28b9o0
To: <sip:901489574204@10.0.1.21>;tag=as2fd54815
Call-ID: 30c12d10-963bc08d@10.0.1.41
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="3352271d",uri="sip:901489574204@10.0.1.21",algorithm=MD5,response="4c430167f5abebad08a371728a5b4c81"
Contact: <sip:3101@10.0.1.41:5060>
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.0.1.40:5061:
OPTIONS sip:10.0.1.40:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK744752d0;rport
From: "asterisk" <sip:asterisk@10.0.1.21>;tag=as15eb1f97
To: <sip:10.0.1.40:5061>
Contact: <sip:asterisk@10.0.1.21>
Call-ID: 6fe3b1f029f2e6766c4063176ea70da1@10.0.1.21
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 21 Dec 2007 13:37:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
akira*CLI>
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 486 Busy Here
To: <sip:10.0.1.40:5061>;tag=b328fe0c274b6cc7i1
From: "asterisk" <sip:asterisk@10.0.1.21>;tag=as15eb1f97
Call-ID: 6fe3b1f029f2e6766c4063176ea70da1@10.0.1.21
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK744752d0
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6fe3b1f029f2e6766c4063176ea70da1@10.0.1.21' Method: OPTIONS
akira*CLI>
<--- SIP read from 10.0.1.41:5060 --->
BYE sip:901489574204@10.0.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-212b23c0
From: <sip:3101@10.0.1.21>;tag=b2137b34817f28b9o0
To: <sip:901489574204@10.0.1.21>;tag=as2fd54815
Call-ID: 30c12d10-963bc08d@10.0.1.41
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="3352271d",uri="sip:901489574204@10.0.1.21",algorithm=MD5,response="8894196a8344d6f39f639cf049b9499a"
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.1.41 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.0.1.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-212b23c0;received=10.0.1.41
From: <sip:3101@10.0.1.21>;tag=b2137b34817f28b9o0
To: <sip:901489574204@10.0.1.21>;tag=as2fd54815
Call-ID: 30c12d10-963bc08d@10.0.1.41
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:901489574204@10.0.1.21>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:901489574204@10.0.1.40:5061> for address/port to send to
set_destination: set destination to 10.0.1.40, port 5061
Reliably Transmitting (no NAT) to 10.0.1.40:5061:
BYE sip:901489574204@10.0.1.40:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK175be556;rport
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as7f28049e
To: <sip:901489574204@10.0.1.40:5061>;tag=52de3dd8aa3e8526i1
Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (handsets, 901489574204, 1) exited non-zero on 'SIP/3101-081e1220'
akira*CLI>
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 200 OK
To: <sip:901489574204@10.0.1.40:5061>;tag=52de3dd8aa3e8526i1
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as7f28049e
Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK175be556
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21' Method: INVITE
Really destroying SIP dialog '30c12d10-963bc08d@10.0.1.41' Method: BYE