rendered paste body<--- SIP read from 10.1.1.201:5060 --->
INVITE sip:912@10.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKcc127a5f8f20fa10
From: Max Nemkov <sip:914@10.1.1.1>;tag=7a170290
To: 912 <sip:912@10.1.1.1>
Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
CSeq: 1 INVITE
Contact: <sip:914@10.1.1.201>
Max-Forwards: 70
Subject: SIP Call
Content-Type: application/sdp
Content-Length: 268
v=0
o=- 423 423 IN IP4 10.1.1.201
s=SIP Session
c=IN IP4 10.1.1.201
t=0 0
m=audio 10000 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=ptime:60
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 13 lines) ---
Sending to 10.1.1.201 : 5060 (no NAT)
Using INVITE request as basis request - 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
<--- Reliably Transmitting (no NAT) to 10.1.1.201:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKcc127a5f8f20fa10;received=10.1.1.201
From: Max Nemkov <sip:914@10.1.1.1>;tag=7a170290
To: 912 <sip:912@10.1.1.1>;tag=as77b85ea7
Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="44ecaec7"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201' in 32000 ms (Method: INVITE)
Found user '914'
ch*CLI>
<--- SIP read from 10.1.1.201:5060 --->
ACK sip:912@10.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKcc127a5f8f20fa10
From: Max Nemkov <sip:914@10.1.1.1>;tag=7a170290
To: 912 <sip:912@10.1.1.1>;tag=as77b85ea7
Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
ch*CLI>
<--- SIP read from 10.1.1.201:5060 --->
INVITE sip:912@10.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK7525cb2ed59b3ed3
From: Max Nemkov <sip:914@10.1.1.1>;tag=7a170290
To: 912 <sip:912@10.1.1.1>
Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
CSeq: 2 INVITE
Contact: <sip:914@10.1.1.201>
Max-Forwards: 70
Subject: SIP Call
Proxy-Authorization: Digest algorithm=MD5,nonce="44ecaec7",realm="asterisk",response="d1b5416c2ee8a222ccd98c65693f9481",uri="sip:912@10.1.1.1:5060",username="914"
Content-Type: application/sdp
Content-Length: 268
v=0
o=- 423 423 IN IP4 10.1.1.201
s=SIP Session
c=IN IP4 10.1.1.201
t=0 0
m=audio 10000 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=ptime:60
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 13 lines) ---
Sending to 10.1.1.201 : 5060 (no NAT)
Using INVITE request as basis request - 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
Found user '914'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.201:10000
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.201:10000
Looking for 912 in internal-phones (domain 10.1.1.1)
list_route: hop: <sip:914@10.1.1.201>
<--- Transmitting (no NAT) to 10.1.1.201:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK7525cb2ed59b3ed3;received=10.1.1.201
From: Max Nemkov <sip:914@10.1.1.1>;tag=7a170290
To: 912 <sip:912@10.1.1.1>
Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:912@10.1.1.1>
Content-Length: 0
<------------>
-- Executing [912@internal-phones:1] NoOp("SIP/914-08691000", "") in new stack
-- Executing [912@internal-phones:2] Dial("SIP/914-08691000", "SIP/sr/912| 30") in new stack
-- Called sr/912
-- SIP/sr-086ae000 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [912@internal-phones:3] Hangup("SIP/914-08691000", "") in new stack
== Spawn extension (internal-phones, 912, 3) exited non-zero on 'SIP/914-08691000'
Scheduling destruction of SIP dialog '79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 10.1.1.201:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK7525cb2ed59b3ed3;received=10.1.1.201
rom: Max Nemkov <sip:914@10.1.1.1>;tag=7a170290
To: 912 <sip:912@10.1.1.1>;tag=as7eef1f27
Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
ch*CLI>
<--- SIP read from 10.1.1.201:5060 --->
ACK sip:912@10.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK7525cb2ed59b3ed3
From: Max Nemkov <sip:914@10.1.1.1>;tag=7a170290
To: 912 <sip:912@10.1.1.1>;tag=as7eef1f27
Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '53e6f47e4dc354155a26c4ec7edee0d6@10.9.0.17' Method: INVITE
ch*CLI>
<--- SIP read from 10.1.1.201:5060 --->
REGISTER sip:10.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKb6071f2c8f04a237
From: 914 <sip:914@10.1.1.1>;tag=492253285abc1822134075
To: 914 <sip:914@10.1.1.1>
Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201
CSeq: 15 REGISTER
Contact: <sip:914@10.1.1.201>
Max-Forwards: 70
Authorization: Digest algorithm=MD5,nonce="3fd19df6",realm="asterisk",response="45e0553004a8f79d449d555d028d6cdf",uri="sip:10.1.1.1:5060",username="914"
Expires: 60
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.201 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.1.1.201:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKb6071f2c8f04a237;received=10.1.1.201
From: 914 <sip:914@10.1.1.1>;tag=492253285abc1822134075
To: 914 <sip:914@10.1.1.1>
Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:914@10.1.1.1>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.1.1.201:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKb6071f2c8f04a237;received=10.1.1.201
From: 914 <sip:914@10.1.1.1>;tag=492253285abc1822134075
To: 914 <sip:914@10.1.1.1>;tag=as4139c298
Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201
CSeq: 15 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f5f35bc"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '5a320266465b32b932cd512702151A1a@10.1.1.201' in 32000 ms (Method: REGISTER)
ch*CLI>
<--- SIP read from 10.1.1.201:5060 --->
REGISTER sip:10.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK5f863e46180cef41
From: 914 <sip:914@10.1.1.1>;tag=492253285abc1822134075
To: 914 <sip:914@10.1.1.1>
Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201
CSeq: 16 REGISTER
Contact: <sip:914@10.1.1.201>
Max-Forwards: 70
Authorization: Digest algorithm=MD5,nonce="6f5f35bc",realm="asterisk",response="464986d7f08c27a429bfb23b4b5ebaee",uri="sip:10.1.1.1:5060",username="914"
Expires: 60
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.201 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.1.1.201:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK5f863e46180cef41;received=10.1.1.201
From: 914 <sip:914@10.1.1.1>;tag=492253285abc1822134075
To: 914 <sip:914@10.1.1.1>
Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201
CSeq: 16 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:914@10.1.1.1>
Content-Length: 0
<------------>
ch*CLI>
<--- Transmitting (no NAT) to 10.1.1.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK5f863e46180cef41;received=10.1.1.201
From: 914 <sip:914@10.1.1.1>;tag=492253285abc1822134075
To: 914 <sip:914@10.1.1.1>;tag=as4139c298
Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201
CSeq: 16 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:914@10.1.1.201>;expires=60
Date: Fri, 07 Mar 2008 07:13:43 GMT
Content-Length: 0
<------------>