rendered paste body<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK23D45EDF
CSeq: 1948 INVITE
To: <sip:9214020@192.168.11.124>
Content-Type: application/sdp
From: <sip:214058@192.168.11.124>;tag=857BEA0
Call-ID: 1088116812@192.168.11.123
Subject: sip:214058@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>
v=0
o=username 0 0 IN IP4 192.168.11.123
s=The Funky Flow
c=IN IP4 192.168.11.123
t=0 0
m=audio 32904 RTP/AVP 8 97 3 0
m=audio 32904 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
<------------->
--- (11 headers 12 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 1088116812@192.168.11.123
<--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK23D45EDF;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=857BEA0
To: <sip:9214020@192.168.11.124>;tag=as66e79e9d
Call-ID: 1088116812@192.168.11.123
CSeq: 1948 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e853eec"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1088116812@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214058'
bee*CLI>
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK23D45EDF
CSeq: 1948 ACK
To: <sip:9214020@192.168.11.124>;tag=as66e79e9d
From: <sip:214058@192.168.11.124>;tag=857BEA0
Call-ID: 1088116812@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>
<------------->
--- (9 headers 0 lines) ---
bee*CLI>
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK575936FF
CSeq: 1949 INVITE
To: <sip:9214020@192.168.11.124>
Proxy-Authorization: Digest username="214058", realm="asterisk", nonce="2e853eec", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="9d896546ecf93a9175e5da73dd30f45b", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214058@192.168.11.124>;tag=857BEA0
Call-ID: 1088116812@192.168.11.123
Subject: sip:214058@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>
v=0
o=username 0 0 IN IP4 192.168.11.123
s=The Funky Flow
c=IN IP4 192.168.11.123
t=0 0
m=audio 32904 RTP/AVP 8 97 3 0
m=audio 32904 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
<------------->
--- (12 headers 12 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 1088116812@192.168.11.123
Found user '214058'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Peer audio RTP is at port 192.168.11.123:32904
Found description format PCMU for ID 0
Found description format GSM for ID 3
Found description format PCMA for ID 8
Found description format iLBC for ID 97
Capabilities: us - 0x8091e (gsm|ulaw|alaw|g726|g729|g726aal2|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.123:32904
Looking for 9214020 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:214058@192.168.11.123;transport=udp>
<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK575936FF;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=857BEA0
To: <sip:9214020@192.168.11.124>
Call-ID: 1088116812@192.168.11.123
CSeq: 1949 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Length: 0
<------------>
Audio is at 192.168.11.124 port 11280
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
INVITE sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK693d1714;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as6e940452
To: <sip:214020@192.168.11.120>
Contact: <sip:214058@192.168.11.124>
Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Jul 2007 17:00:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 356
v=0
o=root 8154 8154 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 11280 RTP/AVP 8 111 112 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Jul 10 21:00:15] WARNING[8996]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/cisco-081edce0(256) to SIP/214058-081e8ea0(8)
bee*CLI>
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK693d1714;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as6e940452
To: <sip:214020@192.168.11.120>;tag=9FE9CD9-1F89
Date: Tue, 10 Jul 2007 17:00:13 GMT
Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.11.120:5060:
ACK sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK693d1714;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as6e940452
To: <sip:214020@192.168.11.120>;tag=9FE9CD9-1F89
Contact: <sip:214058@192.168.11.124>
Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '4acf5b9012422e0332a58f136367fd90@192.168.11.124' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
BYE sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK05708293;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as6e940452
To: <sip:214020@192.168.11.120>;tag=9FE9CD9-1F89
Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK575936FF;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=857BEA0
To: <sip:9214020@192.168.11.124>;tag=as63fc0fa6
Call-ID: 1088116812@192.168.11.123
CSeq: 1949 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Length: 0
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
<------------>
bee*CLI>
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK575936FF
CSeq: 1949 ACK
To: <sip:9214020@192.168.11.124>;tag=as63fc0fa6
From: <sip:214058@192.168.11.124>;tag=857BEA0
Call-ID: 1088116812@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1088116812@192.168.11.123' Method: ACK
bee*CLI>
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK05708293;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as6e940452
To: <sip:214020@192.168.11.120>;tag=9FE9CD9-1F89
Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124
CSeq: 103 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[Jul 10 21:00:15] WARNING[8183]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call '4acf5b9012422e0332a58f136367fd90@192.168.11.124'. Giving up.
Really destroying SIP dialog '4acf5b9012422e0332a58f136367fd90@192.168.11.124' Method: INVITE
Really destroying SIP dialog '778552495@192.168.11.123' Method: REGISTER
Really destroying SIP dialog '1202302177@192.168.11.123' Method: ACK