rendered paste body<--- SIP read from 67.55.159.160:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
<--- SIP read from 192.168.2.5:5064 --->
INVITE sip:17201234567@192.168.2.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bKdc6d822cec6a093c
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>
Contact: <sip:2201@192.168.2.5:5064;transport=udp>
Supported: replaces, timer, path
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11421 INVITE
User-Agent: Grandstream GXP2000 1.1.2.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 347
v=0
o=2201 8000 8000 IN IP4 192.168.2.5
s=SIP Call
c=IN IP4 192.168.2.5
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 99 9 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/16000
a=rtpmap:3 GSM/8000
a=ptime:20
<------------->
--- (13 headers 17 lines) ---
Sending to 192.168.2.5 : 5064 (no NAT)
Using INVITE request as basis request - e08be4896afb71bd@192.168.2.5
<--- Reliably Transmitting (no NAT) to 192.168.2.5:5064 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bKdc6d822cec6a093c;received=192.168.2.5
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>;tag=as6b0a727b
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11421 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71faa331"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'e08be4896afb71bd@192.168.2.5' in 32000 ms (Method: INVITE)
Found user '2201'
<--- SIP read from 192.168.2.5:5064 --->
ACK sip:17201234567@192.168.2.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bKdc6d822cec6a093c
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>;tag=as6b0a727b
Contact: <sip:2201@192.168.2.5:5064;transport=udp>
Supported: path
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11421 ACK
User-Agent: Grandstream GXP2000 1.1.2.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from 192.168.2.5:5064 --->
INVITE sip:17201234567@192.168.2.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>
Contact: <sip:2201@192.168.2.5:5064;transport=udp>
Supported: replaces, timer, path
Proxy-Authorization: Digest username="2201", realm="asterisk", algorithm=MD5, uri="sip:17201234567@192.168.2.201", nonce="71faa331", response="59b1507944c33c5b7b45693723ad1354"
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11422 INVITE
User-Agent: Grandstream GXP2000 1.1.2.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 347
v=0
o=2201 8000 8001 IN IP4 192.168.2.5
s=SIP Call
c=IN IP4 192.168.2.5
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 99 9 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/16000
a=rtpmap:3 GSM/8000
a=ptime:20
<------------->
--- (14 headers 17 lines) ---
Sending to 192.168.2.5 : 5064 (no NAT)
Using INVITE request as basis request - e08be4896afb71bd@192.168.2.5
Found user '2201'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 99
Found RTP audio format 9
Found RTP audio format 3
Peer audio RTP is at port 192.168.2.5:5004
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G723 for ID 4
Found description format G729 for ID 18
Found description format G726-32 for ID 2
Found description format iLBC for ID 99
Found description format G722 for ID 9
Found description format GSM for ID 3
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/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 192.168.2.5:5004
Looking for 17201234567 in home (domain 192.168.2.201)
list_route: hop: <sip:2201@192.168.2.5:5064;transport=udp>
<--- Transmitting (no NAT) to 192.168.2.5:5064 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f;received=192.168.2.5
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11422 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17201234567@192.168.2.201>
Content-Length: 0
<------------>
-- Executing [17201234567@home:1] Dial("SIP/2201-081e77d8", "SIP/lesnet_out/17201234567") in new stack
[Apr 13 12:38:50] WARNING[26210]: chan_sip.c:2737 create_addr: No such host: lesnet_out
Really destroying SIP dialog '6f9e474747a255d45fa8e15d7f500ecf@192.168.2.201' Method: INVITE
[Apr 13 12:38:50] WARNING[26210]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [17201234567@home:2] SayDigits("SIP/2201-081e77d8", "17201234567") in new stack
Audio is at 192.168.2.201 port 17244
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
<--- Transmitting (no NAT) to 192.168.2.5:5064 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f;received=192.168.2.5
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>;tag=as6cf07fbc
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11422 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17201234567@192.168.2.201>
Content-Type: application/sdp
Content-Length: 233
v=0
o=root 26180 26180 IN IP4 192.168.2.201
s=session
c=IN IP4 192.168.2.201
t=0 0
m=audio 17244 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- <SIP/2201-081e77d8> Playing 'digits/1' (language 'en')
-- <SIP/2201-081e77d8> Playing 'digits/7' (language 'en')
Really destroying SIP dialog '6929d341645213f668a2f842313f8116@192.168.2.201' Method: REGISTER
Really destroying SIP dialog '1f88d90b32e157b5478dbc2a41e99194@192.168.2.201' Method: REGISTER
-- <SIP/2201-081e77d8> Playing 'digits/2' (language 'en')
-- <SIP/2201-081e77d8> Playing 'digits/0' (language 'en')
-- <SIP/2201-081e77d8> Playing 'digits/9' (language 'en')
-- <SIP/2201-081e77d8> Playing 'digits/3' (language 'en')
-- <SIP/2201-081e77d8> Playing 'digits/8' (language 'en')
<--- SIP read from 192.168.2.5:5064 --->
CANCEL sip:17201234567@192.168.2.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>
Supported: path
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11422 CANCEL
User-Agent: Grandstream GXP2000 1.1.2.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.2.5 : 5064 (no NAT)
<--- Reliably Transmitting (no NAT) to 192.168.2.5:5064 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f;received=192.168.2.5
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>;tag=as6cf07fbc
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11422 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.2.5:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f;received=192.168.2.5
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>;tag=as6cf07fbc
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11422 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17201234567@192.168.2.201>
Content-Length: 0
<------------>
== Spawn extension (home, 17201234567, 2) exited non-zero on 'SIP/2201-081e77d8'
[Apr 13 12:38:55] ERROR[26210]: cdr_addon_mysql.c:148 mysql_log: cdr_mysql: cannot connect to database server localhost.
<--- SIP read from 192.168.2.5:5064 --->
ACK sip:17201234567@192.168.2.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f
From: "Ext 2201" <sip:2201@192.168.2.201>;tag=a812cbed43f79660
To: <sip:17201234567@192.168.2.201>;tag=as6cf07fbc
Contact: <sip:2201@192.168.2.5:5064;transport=udp>
Supported: path
Proxy-Authorization: Digest username="2201", realm="asterisk", algorithm=MD5, uri="sip:17201234567@192.168.2.201", nonce="71faa331", response="59b1507944c33c5b7b45693723ad1354"
Call-ID: e08be4896afb71bd@192.168.2.5
CSeq: 11422 ACK
User-Agent: Grandstream GXP2000 1.1.2.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog 'e08be4896afb71bd@192.168.2.5' Method: ACK
<--- SIP read from 67.55.159.160:5060 --->
<------------->