rendered paste body<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9722490@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK3F982AF1
CSeq: 6021 INVITE
To: <sip:9722490@192.168.11.124>
Content-Type: application/sdp
From: <sip:214059@192.168.11.124>;tag=2B2117DC
Call-ID: 1246610069@192.168.11.123
Subject: sip:214059@192.168.11.124
Content-Length: 232
User-Agent: kphone/4.2
Contact: <sip:214059@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 32840 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 - 1246610069@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=z9hG4bK3F982AF1;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=2B2117DC
To: <sip:9722490@192.168.11.124>;tag=as1b0c06c3
Call-ID: 1246610069@192.168.11.123
CSeq: 6021 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56209a8e"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1246610069@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214059'
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9722490@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK3F982AF1
CSeq: 6021 ACK
To: <sip:9722490@192.168.11.124>;tag=as1b0c06c3
From: <sip:214059@192.168.11.124>;tag=2B2117DC
Call-ID: 1246610069@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214059@192.168.11.123;transport=udp>
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9722490@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4D37957B
CSeq: 6022 INVITE
To: <sip:9722490@192.168.11.124>
Proxy-Authorization: Digest username="214059", realm="asterisk", nonce="56209a8e", uri="sip:9722490@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="2b5c67b4ccb947e1a79c0408f2035eae", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214059@192.168.11.124>;tag=2B2117DC
Call-ID: 1246610069@192.168.11.123
Subject: sip:214059@192.168.11.124
Content-Length: 232
User-Agent: kphone/4.2
Contact: <sip:214059@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 32840 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 - 1246610069@192.168.11.123
Found user '214059'
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 192.168.11.123:32840
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 192.168.11.123:32840
Looking for 9722490 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:214059@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=z9hG4bK4D37957B;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=2B2117DC
To: <sip:9722490@192.168.11.124>
Call-ID: 1246610069@192.168.11.123
CSeq: 6022 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9722490@192.168.11.124>
Content-Length: 0
<------------>
-- Executing [9722490@local-phones:1] Dial("SIP/214059-08208090", "SIP/722490@cisco|20") in new stack
Audio is at 192.168.11.124 port 13740
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:722490@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK6692ff3f;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as4571c879
To: <sip:722490@192.168.11.120>
Contact: <sip:214059@192.168.11.124>
Call-ID: 5f77a2a00e63b7cf64d1d7c4137a7bbd@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 09 Jul 2007 11:41:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 19181 19181 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 13740 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 722490@cisco
[Jul 9 15:41:55] WARNING[19237]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/cisco-082127a0(1) to SIP/214059-08208090(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=z9hG4bK6692ff3f;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as4571c879
To: <sip:722490@192.168.11.120>;tag=3B4D441-6D3
Date: Mon, 09 Jul 2007 11:41:53 GMT
Call-ID: 5f77a2a00e63b7cf64d1d7c4137a7bbd@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=z9hG4bK6692ff3f;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as4571c879
To: <sip:722490@192.168.11.120>;tag=3B4D441-6D3
Date: Mon, 09 Jul 2007 11:41:53 GMT
Call-ID: 5f77a2a00e63b7cf64d1d7c4137a7bbd@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:722490@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK6692ff3f;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as4571c879
To: <sip:722490@192.168.11.120>;tag=3B4D441-6D3
Contact: <sip:214059@192.168.11.124>
Call-ID: 5f77a2a00e63b7cf64d1d7c4137a7bbd@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/cisco-082127a0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [9722490@local-phones:2] Congestion("SIP/214059-08208090", "") 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=z9hG4bK4D37957B;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=2B2117DC
To: <sip:9722490@192.168.11.124>;tag=as42a1bf30
Call-ID: 1246610069@192.168.11.123
CSeq: 6022 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9722490@192.168.11.124>
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
<------------>
== Spawn extension (local-phones, 9722490, 2) exited non-zero on 'SIP/214059-08208090'
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9722490@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4D37957B
CSeq: 6022 ACK
To: <sip:9722490@192.168.11.124>;tag=as42a1bf30
From: <sip:214059@192.168.11.124>;tag=2B2117DC
Call-ID: 1246610069@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214059@192.168.11.123;transport=udp>
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '5f77a2a00e63b7cf64d1d7c4137a7bbd@192.168.11.124' Method: INVITE
Really destroying SIP dialog '1246610069@192.168.11.123' Method: ACK