<--- SIP read from UDP:10.10.19.28:54113 --->
INVITE sip:999017@10.10.19.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.28:5060;x-route-tag="cid:15817@10.10.19.28";branch=z9hG4bK1DA6F6
Remote-Party-ID: <sip:2000@10.10.19.28>;party=calling;screen=no;privacy=off
From: <sip:2000@10.10.19.28>;tag=33F281F0-1DFB
To: <sip:999017@10.10.19.3>
Date: Wed, 12 Feb 2014 16:19:27 GMT
Call-ID: 477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1199460414-2469925347-2160491625-4139583488
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1392221967
Contact: <sip:2000@10.10.19.28:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 270

v=0
o=CiscoSystemsSIP-GW-UserAgent 2628 4435 IN IP4 10.10.19.28
s=SIP Call
c=IN IP4 10.10.19.28
t=0 0
m=audio 16752 RTP/AVP 18 101
c=IN IP4 10.10.19.28
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:60
<------------->
--- (21 headers 12 lines) ---
Sending to 10.10.19.28:5060 (no NAT)
Sending to 10.10.19.28:5060 (no NAT)
Using INVITE request as basis request - 477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28
No matching peer for '2000' from '10.10.19.28:54113'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.19.28:16752
Looking for 999017 in default (domain 10.10.19.3)
list_route: hop: <sip:2000@10.10.19.28:5060>

<--- Transmitting (no NAT) to 10.10.19.28:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.19.28:5060;x-route-tag="cid:15817@10.10.19.28";branch=z9hG4bK1DA6F6;received=10.10.19.28
From: <sip:2000@10.10.19.28>;tag=33F281F0-1DFB
To: <sip:999017@10.10.19.3>
Call-ID: 477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28
CSeq: 101 INVITE
Server: Asterisk Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:999017@10.10.19.3:5060>
Content-Length: 0






<------------>
    -- Executing [999017@default:1] Dial("SIP/10.10.19.28-00000014", "SIP/5555@10.10.4.58,,e") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 14592
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.4.58:5060:
INVITE sip:5555@10.10.4.58 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK13200307
Max-Forwards: 70
From: <sip:2000@10.10.19.3>;tag=as3911212c
To: <sip:5555@10.10.4.58>
Contact: <sip:2000@10.10.19.3:5060>
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
CSeq: 102 INVITE
User-Agent: Asterisk Server
Date: Wed, 12 Feb 2014 16:19:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 918198641 918198641 IN IP4 10.10.19.3
s=Asterisk PBX 11.7.0
c=IN IP4 10.10.19.3
t=0 0
m=audio 14592 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv






---
    -- Called SIP/5555@10.10.4.58
Retransmitting #1 (no NAT) to 10.10.4.58:5060:
INVITE sip:5555@10.10.4.58 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK13200307
Max-Forwards: 70
From: <sip:2000@10.10.19.3>;tag=as3911212c
To: <sip:5555@10.10.4.58>
Contact: <sip:2000@10.10.19.3:5060>
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
CSeq: 102 INVITE
User-Agent: Asterisk Server
Date: Wed, 12 Feb 2014 16:19:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 918198641 918198641 IN IP4 10.10.19.3
s=Asterisk PBX 11.7.0
c=IN IP4 10.10.19.3
t=0 0
m=audio 14592 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.10.4.56:5060 --->
SIP/2.0 100 Trying
Date: Wed, 12 Feb 2014 16:20:12 GMT
From: <sip:2000@10.10.19.3>;tag=as3911212c
Allow-Events: telephone-event
Content-Length: 0
To: <sip:5555@10.10.4.58>
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK13200307
CSeq: 102 INVITE
Server: Cisco-SIPGateway/IOS-12.x

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:10.10.4.56:5060 --->
SIP/2.0 200 OK
Date: Wed, 12 Feb 2014 16:20:12 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:2000@10.10.19.3>;tag=as3911212c
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Content-Length: 267
To: <sip:5555@10.10.4.58>;tag=888AC2DC-1FEC
Contact: <sip:5555@10.10.4.56:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK13200307
CSeq: 102 INVITE
Server: Cisco-SIPGateway/IOS-12.x

v=0
o=CiscoSystemsSIP-GW-UserAgent 5853 2684 IN IP4 10.10.4.56
s=SIP Call
c=IN IP4 10.10.4.56
t=0 0
m=audio 17366 RTP/AVP 18 101
c=IN IP4 10.10.4.56
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (16 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.4.56:17366
list_route: hop: <sip:5555@10.10.4.56:5060>
set_destination: Parsing <sip:5555@10.10.4.56:5060> for address/port to send to
set_destination: set destination to 10.10.4.56:5060
Transmitting (no NAT) to 10.10.4.56:5060:
ACK sip:5555@10.10.4.56:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK3d5e36de
Max-Forwards: 70
From: <sip:2000@10.10.19.3>;tag=as3911212c
To: <sip:5555@10.10.4.58>;tag=888AC2DC-1FEC
Contact: <sip:2000@10.10.19.3:5060>
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
CSeq: 102 ACK
User-Agent: Asterisk Server
Content-Length: 0


---
    -- SIP/10.10.4.58-00000015 answered SIP/10.10.19.28-00000014
Audio is at 13052
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.10.19.28:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.19.28:5060;x-route-tag="cid:15817@10.10.19.28";branch=z9hG4bK1DA6F6;received=10.10.19.28
From: <sip:2000@10.10.19.28>;tag=33F281F0-1DFB
To: <sip:999017@10.10.19.3>;tag=as215096a3
Call-ID: 477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28
CSeq: 101 INVITE
Server: Asterisk Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:999017@10.10.19.3:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 283

v=0
o=root 1631856383 1631856383 IN IP4 10.10.19.3
s=Asterisk PBX 11.7.0
c=IN IP4 10.10.19.3
t=0 0
m=audio 13052 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:60
a=sendrecv

<------------>

<--- SIP read from UDP:10.10.19.28:54113 --->
ACK sip:999017@10.10.19.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.28:5060;x-route-tag="cid:15817@10.10.19.28";branch=z9hG4bK1DBB1F
From: <sip:2000@10.10.19.28>;tag=33F281F0-1DFB
To: <sip:999017@10.10.19.3>;tag=as215096a3
Date: Wed, 12 Feb 2014 16:19:27 GMT
Call-ID: 477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
       > 0x7fd104026520 -- Probation passed - setting RTP source address to 10.10.19.28:16752
       > 0x7fd0f4009600 -- Probation passed - setting RTP source address to 10.10.4.56:17366

<--- SIP read from UDP:10.10.4.56:5060 --->
SIP/2.0 200 OK
Date: Wed, 12 Feb 2014 16:20:12 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:2000@10.10.19.3>;tag=as3911212c
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Content-Length: 267
To: <sip:5555@10.10.4.58>;tag=888AC2DC-1FEC
Contact: <sip:5555@10.10.4.56:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK13200307
CSeq: 102 INVITE
Server: Cisco-SIPGateway/IOS-12.x

v=0
o=CiscoSystemsSIP-GW-UserAgent 5853 2684 IN IP4 10.10.4.56
s=SIP Call
c=IN IP4 10.10.4.56
t=0 0
m=audio 17366 RTP/AVP 18 101
c=IN IP4 10.10.4.56
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (16 headers 12 lines) ---
set_destination: Parsing <sip:5555@10.10.4.56:5060> for address/port to send to
set_destination: set destination to 10.10.4.56:5060
Transmitting (no NAT) to 10.10.4.56:5060:
ACK sip:5555@10.10.4.56:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK3d1f1b98
Max-Forwards: 70
From: <sip:2000@10.10.19.3>;tag=as3911212c
To: <sip:5555@10.10.4.58>;tag=888AC2DC-1FEC
Contact: <sip:2000@10.10.19.3:5060>
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
CSeq: 102 ACK
User-Agent: Asterisk Server
Content-Length: 0


---
    -- Remote UNIX connection

<--- SIP read from UDP:10.10.19.28:54113 --->
BYE sip:999017@10.10.19.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.28:5060;x-route-tag="cid:15817@10.10.19.28";branch=z9hG4bK1DC1353
From: <sip:2000@10.10.19.28>;tag=33F281F0-1DFB
To: <sip:999017@10.10.19.3>;tag=as215096a3
Date: Wed, 12 Feb 2014 16:19:27 GMT
Call-ID: 477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Max-Forwards: 70
Timestamp: 1392221995
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=452,OS=27120,PR=160,OR=9600,PL=0,JI=16,LA=1,DU=27
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 10.10.19.28:5060 (no NAT)
Scheduling destruction of SIP dialog '477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.10.19.28:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.19.28:5060;x-route-tag="cid:15817@10.10.19.28";branch=z9hG4bK1DC1353;received=10.10.19.28
From: <sip:2000@10.10.19.28>;tag=33F281F0-1DFB
To: <sip:999017@10.10.19.3>;tag=as215096a3
Call-ID: 477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28
CSeq: 102 BYE
Server: Asterisk Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:5555@10.10.4.56:5060> for address/port to send to
set_destination: set destination to 10.10.4.56:5060
Reliably Transmitting (no NAT) to 10.10.4.56:5060:
BYE sip:5555@10.10.4.56:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK35f50b54
Max-Forwards: 70
From: <sip:2000@10.10.19.3>;tag=as3911212c
To: <sip:5555@10.10.4.58>;tag=888AC2DC-1FEC
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
CSeq: 103 BYE
User-Agent: Asterisk Server
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (default, 999017, 1) exited non-zero on 'SIP/10.10.19.28-00000014'
Retransmitting #1 (no NAT) to 10.10.4.56:5060:
BYE sip:5555@10.10.4.56:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK35f50b54
Max-Forwards: 70
From: <sip:2000@10.10.19.3>;tag=as3911212c
To: <sip:5555@10.10.4.58>;tag=888AC2DC-1FEC
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
CSeq: 103 BYE
User-Agent: Asterisk Server
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:10.10.4.56:5060 --->
SIP/2.0 200 OK
Reason: Q.850;cause=16
Date: Wed, 12 Feb 2014 16:20:40 GMT
From: <sip:2000@10.10.19.3>;tag=as3911212c
Content-Length: 0
To: <sip:5555@10.10.4.58>;tag=888AC2DC-1FEC
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK35f50b54
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 BYE

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060' Method: INVITE

<--- SIP read from UDP:10.10.4.56:5060 --->
SIP/2.0 200 OK
From: <sip:2000@10.10.19.3>;tag=as3911212c
Content-Length: 0
To: <sip:5555@10.10.4.58>;tag=888AC2DC-1FEC
Call-ID: 620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060
Via: SIP/2.0/UDP 10.10.19.3:5060;branch=z9hG4bK35f50b54
CSeq: 103 BYE

<------------->
--- (7 headers 0 lines) ---
    -- Remote UNIX connection disconnected
Really destroying SIP dialog '477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28' Method: BYE
    -- Remote UNIX connection
hcc-dev1-avp*CLI> sip show peer peerHSRP


  * Name       : peerHSRP
  Description  :
  Realtime peer: No
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : default
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     :
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Max forwards : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Force rport  : Auto (No)
  Symmetric RTP: No
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 10.10.4.58
  Addr->IP     : 10.10.4.58:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username:
  SIP Options  : (none)
  Codecs       : (g729)
  Codec Order  : (g729:60)
  Auto-Framing :  No
  Status       : Unmonitored
  Useragent    :
  Reg. Contact :
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

hcc-dev1-avp*CLI>
hcc-dev1-avp*CLI> core show channel SIP/10.10.4.58-00000015
 -- General --
           Name: SIP/10.10.4.58-00000015
           Type: SIP
       UniqueID: 1392221967.21
       LinkedID: 1392221967.20
      Caller ID: 999017
 Caller ID Name: (N/A)
Connected Line ID: 2000
Connected Line ID Name: (N/A)
Eff. Connected Line ID: 2000
Eff. Connected Line ID Name: (N/A)
    DNID Digits: (N/A)
       Language: en
          State: Up (6)
          Rings: 0
  NativeFormats: (g729)
    WriteFormat: g729
     ReadFormat: g729
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: 28
      Frames in: 165
     Frames out: 390
 Time to Hangup: 0
   Elapsed Time: 0h0m24s
  Direct Bridge: SIP/10.10.19.28-00000014
Indirect Bridge: SIP/10.10.19.28-00000014
 --   PBX   --
        Context: default
      Extension:
       Priority: 1
     Call Group: 0
   Pickup Group: 0
    Application: AppDial
           Data: (Outgoing Line)
    Blocking in: ast_waitfor_nandfds
 Call Identifer: [C-0000000a]
      Variables:
BRIDGEPVTCALLID=477EEC66-933811E3-8487D367-DBF5FAC4@10.10.19.28
BRIDGEPEER=SIP/10.10.19.28-00000014
DIALEDPEERNUMBER=5555@10.10.4.58
SIPCALLID=620ace9467aa3017090f98a54491d8e8@10.10.19.3:5060

  CDR Variables:
level 1: dnid=
level 1: clid=999017
level 1: src=999017
level 1: dst=s
level 1: dcontext=default
level 1: channel=SIP/10.10.4.58-00000015
level 1: start=2014-02-12 16:19:27
level 1: answer=2014-02-12 16:19:28
level 1: duration=24
level 1: billsec=23
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
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