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=z9hG4bK45EDE10
CSeq: 5236 INVITE
To: <sip:9214020@192.168.11.124>
Content-Type: application/sdp
From: <sip:999777@192.168.11.124>;tag=746182A9
Call-ID: 2130229940@192.168.11.123
Subject: sip:999777@192.168.11.124
Content-Length: 232
User-Agent: kphone/4.2
Contact: <sip:999777@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 32816 RTP/AVP 0 97 8 3
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 11 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 2130229940@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=z9hG4bK45EDE10;received=192.168.11.123
From: <sip:999777@192.168.11.124>;tag=746182A9
To: <sip:9214020@192.168.11.124>;tag=as0665fd76
Call-ID: 2130229940@192.168.11.123
CSeq: 5236 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="14eb84fb"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '2130229940@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '999777'
<--- 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=z9hG4bK45EDE10
CSeq: 5236 ACK
To: <sip:9214020@192.168.11.124>;tag=as0665fd76
From: <sip:999777@192.168.11.124>;tag=746182A9
Call-ID: 2130229940@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:999777@192.168.11.123;transport=udp>
<------------->
--- (9 headers 0 lines) ---
<--- 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=z9hG4bK73C42E7D
CSeq: 5237 INVITE
To: <sip:9214020@192.168.11.124>
Proxy-Authorization: Digest username="999777", realm="asterisk", nonce="14eb84fb", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="00fa762d8827bcb6426e1e1e832baf65", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:999777@192.168.11.124>;tag=746182A9
Call-ID: 2130229940@192.168.11.123
Subject: sip:999777@192.168.11.124
Content-Length: 232
User-Agent: kphone/4.2
Contact: <sip:999777@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 32816 RTP/AVP 0 97 8 3
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 11 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 2130229940@192.168.11.123
Found user '999777'
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 3
Peer audio RTP is at port 192.168.11.123:32816
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 - 0x8000e (gsm|ulaw|alaw|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 192.168.11.123:32816
Looking for 9214020 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:999777@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=z9hG4bK73C42E7D;received=192.168.11.123
From: <sip:999777@192.168.11.124>;tag=746182A9
To: <sip:9214020@192.168.11.124>
Call-ID: 2130229940@192.168.11.123
CSeq: 5237 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
<------------>
-- Executing [9214020@local-phones:1] Dial("SIP/999777-08208050", "SIP/214020@cisco|20") in new stack
Audio is at 192.168.11.124 port 16402
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) 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=z9hG4bK32660968;rport
From: "999777" <sip:999777@192.168.11.124>;tag=as516a504f
To: <sip:214020@192.168.11.120>
Contact: <sip:999777@192.168.11.124>
Call-ID: 7f96487a0167b0691131d51f422125f1@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 09 Jul 2007 11:29:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 19057 19057 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 16402 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
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 214020@cisco
[Jul 9 15:29:44] WARNING[19133]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/cisco-08212628(1) to SIP/999777-08208050(4)
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK32660968;rport
From: "999777" <sip:999777@192.168.11.124>;tag=as516a504f
To: <sip:214020@192.168.11.120>;tag=3A9AB20-1C8E
Date: Mon, 09 Jul 2007 11:29:42 GMT
Call-ID: 7f96487a0167b0691131d51f422125f1@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK32660968;rport
From: "999777" <sip:999777@192.168.11.124>;tag=as516a504f
To: <sip:214020@192.168.11.120>;tag=3A9AB20-1C8E
Date: Mon, 09 Jul 2007 11:29:42 GMT
Call-ID: 7f96487a0167b0691131d51f422125f1@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=z9hG4bK32660968;rport
From: "999777" <sip:999777@192.168.11.124>;tag=as516a504f
To: <sip:214020@192.168.11.120>;tag=3A9AB20-1C8E
Contact: <sip:999777@192.168.11.124>
Call-ID: 7f96487a0167b0691131d51f422125f1@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/cisco-08212628 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [9214020@local-phones:2] Congestion("SIP/999777-08208050", "") in new stack
<--- 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=z9hG4bK73C42E7D;received=192.168.11.123
From: <sip:999777@192.168.11.124>;tag=746182A9
To: <sip:9214020@192.168.11.124>;tag=as4400d7a0
Call-ID: 2130229940@192.168.11.123
CSeq: 5237 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: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
<------------>
== Spawn extension (local-phones, 9214020, 2) exited non-zero on 'SIP/999777-08208050'
<--- 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=z9hG4bK73C42E7D
CSeq: 5237 ACK
To: <sip:9214020@192.168.11.124>;tag=as4400d7a0
From: <sip:999777@192.168.11.124>;tag=746182A9
Call-ID: 2130229940@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:999777@192.168.11.123;transport=udp>
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '7f96487a0167b0691131d51f422125f1@192.168.11.124' Method: INVITE
Really destroying SIP dialog '2130229940@192.168.11.123' Method: ACK