Audio stops after long playback interval

Asterisk Asterisk Support
1

We are Using The Asterisk® 1.6.2.6 with konference 1.4.
I have one hardfone using pjsua 1.8 as the useragent… (sip:0001)transmitting audio to the Konference ID(0901).The above Konference has other receivers (also same user agents as the transmitter eg sip:0110,sip:0112…etc).
This scheme of things runs fine for some hours, upon a certain elapsed time something like a change in day the audio is no longer heard on the receiver even if all transmitter and receiver are registered at asterisk and sip channels are active.
Please find attached log from asterisk debug channels. How do I proceed to debug this issue?

Asterisk 1.6.2.6, Copyright (C) 1999 - 2010 Digium, Inc. and others.
=========================================================================
Connected to Asterisk 1.6.2.6 currently running on node2 (pid = 2963)
node2*CLI> 
Verbosity is at least 3

  quit    sip show peer 0001
  * Name       : 0001
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : users
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 205
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  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       : 
  Addr->IP     : 192.168.0.14 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 0001
  SIP Options  : (none)
  Codecs       : 0x1000 (g722)
  Codec Order  : (g722:20)
  Auto-Framing :  No 
  100 on REG   : No
  Status       : Unmonitored
  Useragent    : PJSUA v1.8/arm-none-linux-gnueabi
  Reg. Contact : sip:0001@192.168.0.14:5060
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uac
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  Parkinglot   : 

  sip show peer 0001 
  * Name       : 0110
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : users
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 201
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  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       : 
  Addr->IP     : 192.168.0.10 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 0110
  SIP Options  : (none)
  Codecs       : 0x1000 (g722)
  Codec Order  : (g722:20)
  Auto-Framing :  No 
  100 on REG   : No
  Status       : Unmonitored
  Useragent    : PJSUA v1.8/arm-none-linux-gnueabi
  Reg. Contact : sip:0110@192.168.0.10:5060
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  Parkinglot   : 


  core show channels
Channel              Location             State   Application(Data)             
SIP/0110-00000019    0901@users:2         Up      Konference(0901)              
SIP/0001-00000016    0901@users:2         Up      Konference(0901)              
2 active channels
2 active calls
12 calls processed

  core show channels 
Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry
192.168.0.14     0001             tw-DEhUyx-Vude5  0x1000 (g722)    No       Rx: ACK                   
192.168.0.10     0110             5725813e69ce4ae  0x1000 (g722)    No       Tx: ACK                   
2 active SIP dialogs

  sip  show channels   tw-DEhUyx-Vude5GEf-fynbATF3pQ25D 
  * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                tw-DEhUyx-Vude5GEf-fynbATF3pQ25D
  Owner channel ID:       SIP/0001-00000016
  Our Codec Capability:   4096
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   4110
  Joint Codec Capability:   4096
  Format:                 0x1000 (g722)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.0.14:5060
  Received Address:       192.168.0.14:5060
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               192.168.0.3 (local)
  Our Tag:                as65db3dee
  Their Tag:              TnngYjrETLOyQYXThpt.CnlFC4MxIhkm
  SIP User agent:         PJSUA v1.8/arm-none-linux-gnueabi
  Username:               0001
  Peername:               0001
  Original uri:           sip:0001@192.168.0.14:5060
  Caller-ID:              0001
  Need Destroy:           No
  Last Message:           Rx: ACK
  Promiscuous Redir:      No
  Route:                  sip:0001@192.168.0.14:5060
  DTMF Mode:              rfc2833
  SIP Options:            100rel norefersub replaces replace timer 
  Session-Timer:          Active
  S-Timer Interval:       86400
  S-Timer Refresher:      uac
  S-Timer Expirys:        0
  S-Timer Sched Id:       245
  S-Timer Peer Sts:       Active
  S-Timer Cached Min-SE:  90
  S-Timer Cached SE:      1800
  S-Timer Cached Ref:     uac
  S-Timer Cached Mode:    Accept

  


  sip  show channel 5725813e69ce4ae26da4e6ad76eade69@192.168.0.3
  * SIP Call
  Curr. trans. direction:  Outgoing
  Call-ID:                5725813e69ce4ae26da4e6ad76eade69@192.168.0.3
  Owner channel ID:       SIP/0110-00000019
  Our Codec Capability:   4096
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   4096
  Joint Codec Capability:   4096
  Format:                 0x1000 (g722)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.0.10:5060
  Received Address:       192.168.0.10:5060
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               192.168.0.3 (local)
  Our Tag:                as685ebb31
  Their Tag:              8DEMCabPHdA1TWmYQBKPIV8awP1lDic9
  SIP User agent:         
  Username:               0110
  Peername:               0110
  Original uri:           sip:0110@192.168.0.10:5060
  Need Destroy:           No
  Last Message:           Tx: ACK
  Promiscuous Redir:      No
  Route:                  sip:0110@192.168.0.10:5060
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive

  core  show channel SIP/0001-00000016 
node2*CLI> 
 -- General --
           Name: SIP/0001-00000016
           Type: SIP
       UniqueID: 1343933224.23
      Caller ID: 0001
 Caller ID Name: (N/A)
    DNID Digits: 0901
       Language: en
          State: Up (6)
          Rings: 0
  NativeFormats: 0x1000 (g722)
    WriteFormat: 0x1000 (g722)
     ReadFormat: 0x1000 (g722)
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: 28
      Frames in: 22947
     Frames out: 22892
 Time to Hangup: 0
   Elapsed Time: 0h7m38s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: users
      Extension: 0901
       Priority: 2
     Call Group: 0
   Pickup Group: 0
    Application: Konference
           Data: 0901
    Blocking in: ast_waitfor_nandfds
      Variables:
SIPCALLID=tw-DEhUyx-Vude5GEf-fynbATF3pQ25D
SIPDOMAIN=192.168.0.3
SIPURI=sip:0001@192.168.0.14:5060

  CDR Variables:
level 1: dnid=0901
level 1: clid=0001
level 1: src=0001
level 1: dst=0901
level 1: dcontext=users
level 1: channel=SIP/0001-00000016
level 1: lastapp=Konference
level 1: lastdata=0901
level 1: start=2012-08-03 00:17:04
level 1: answer=2012-08-03 00:17:04
level 1: duration=457
level 1: billsec=457
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1343933224.23



  core  show channel SIP/0110-00000019 
node2*CLI> 
 -- General --
           Name: SIP/0110-00000019
           Type: SIP
       UniqueID: 1343933224.26
      Caller ID: 0901
 Caller ID Name: (N/A)
    DNID Digits: (N/A)
       Language: en
          State: Up (6)
          Rings: 0
  NativeFormats: 0x1000 (g722)
    WriteFormat: 0x1000 (g722)
     ReadFormat: 0x1000 (g722)
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: 46
      Frames in: 22992
     Frames out: 24514
 Time to Hangup: 0
   Elapsed Time: 0h8m10s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: users
      Extension: 0901
       Priority: 2
     Call Group: 0
   Pickup Group: 0
    Application: Konference
           Data: 0901
    Blocking in: ast_waitfor_nandfds

  
      Variables:
host=dynamic
type=friend
quality=no
secret=0110
SIPCALLID=5725813e69ce4ae26da4e6ad76eade69@192.168.0.3


  
  CDR Variables:
level 1: clid=0901
level 1: src=0901
level 1: dst=0901
level 1: dcontext=users
level 1: channel=SIP/0110-00000019
level 1: lastapp=Konference
level 1: lastdata=0901
level 1: start=2012-08-03 00:17:04
level 1: duration=490
level 1: billsec=0
level 1: disposition=NO ANSWER
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1343933224.26


  core  show channel SIP/0110-00000019 [12G [K  core  show channel SIP/0110-00000019 [36G001-00000016 [36G110-00000019 [12G [K  asterisk  
  == Using SIP RTP CoS mark 5

  asterisk
    -- Got SIP response 486 "Busy Here" back from 192.168.0.14

  asterisk
  == Using SIP RTP CoS mark 5

  asterisk
    -- Got SIP response 486 "Busy Here" back from 192.168.0.13

  asterisk
  == Using SIP RTP CoS mark 5

  asterisk
    -- Got SIP response 486 "Busy Here" back from 192.168.0.10

  asterisk
  == Using SIP RTP CoS mark 5

  asterisk
    -- Got SIP response 486 "Busy Here" back from 192.168.0.22

  asterisk
  == Using SIP RTP CoS mark 5

  asterisk
    -- Got SIP response 486 "Busy Here" back from 192.168.1.15

core set debug 9
node2*CLI> 
Core debug was 0 and is now 9
<--- SIP read from UDP:192.168.0.14:5060 --->


<------------->

  sip set debug ip 192.168.0.14  [K  [K10
node2*CLI> 
SIP Debugging Enabled for IP: 192.168.0.10

  


<--- SIP read from UDP:192.168.0.14:5060 --->


<------------->

 
  sip set debug peer 0001
node2*CLI> 
SIP Debugging Enabled for IP: 192.168.0.14:5060

  sip set debug peer 0001
<--- SIP read from UDP:192.168.0.14:5060 --->


<------------->

  sip set debug peer 0001  
node2*CLI> 
SIP Debugging Enabled for IP: 192.168.0.10:5060

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjcKeHW9DuAmpMBnN4jaGaaYcqIFESnTwJ
Max-Forwards: 70
From: <sip:0010@192.168.0.3>;tag=M2vHPmjdzUNs93KW5pXTQP7.mmMnHfUx
To: <sip:0010@192.168.0.3>
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20806 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0010@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjcKeHW9DuAmpMBnN4jaGaaYcqIFESnTwJ;received=192.168.0.10;rport=5060
From: <sip:0010@192.168.0.3>;tag=M2vHPmjdzUNs93KW5pXTQP7.mmMnHfUx
To: <sip:0010@192.168.0.3>;tag=as4f0eff66
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20806 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1886a43d"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' in 32000 ms (Method: REGISTER)

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjEkfHbNmKixv34VND7M2qu5s7wS.ZNG4K
Max-Forwards: 70
From: <sip:0010@192.168.0.3>;tag=M2vHPmjdzUNs93KW5pXTQP7.mmMnHfUx
To: <sip:0010@192.168.0.3>
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20807 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0010@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0010", realm="asterisk", nonce="1886a43d", uri="sip:192.168.0.3", response="f5ffcd2ba3cab9accc20704be681e5fb", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

  
<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjEkfHbNmKixv34VND7M2qu5s7wS.ZNG4K;received=192.168.0.10;rport=5060
From: <sip:0010@192.168.0.3>;tag=M2vHPmjdzUNs93KW5pXTQP7.mmMnHfUx
To: <sip:0010@192.168.0.3>;tag=as4f0eff66
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20807 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
C
  
ontact: <sip:0010@192.168.0.10:5060>;expires=300
Date: Thu, 02 Aug 2012 19:00:40 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' in 32000 ms (Method: REGISTER)

  
  == Using SIP RTP CoS mark 5

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPj-VMeQLuryMAk3kJ3gAyQj5s2judznqB5
Max-Forwards: 70
From: <sip:0110@192.168.0.3>;tag=aHgaBxKFqtTjzfv6BJWBFE72SRB-TKQo
To: <sip:0110@192.168.0.3>
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23501 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0110@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPj-VMeQLuryMAk3kJ3gAyQj5s2judznqB5;received=192.168.0.10;rport=5060
From: <sip:0110@192.168.0.3>;tag=aHgaBxKFqtTjzfv6BJWBFE72SRB-TKQo
To: <sip:0110@192.168.0.3>;tag=as6ddcbf4e
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23501 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ad7d217"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' in 32000 ms (Method: REGISTER)

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.14

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjEC0VOFXKTCu9tih.uD7abn6xfbcJ5tIM
Max-Forwards: 70
From: <sip:0110@192.168.0.3>;tag=aHgaBxKFqtTjzfv6BJWBFE72SRB-TKQo
To: <sip:0110@192.168.0.3>
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23502 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0110@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0110", realm="asterisk", nonce="2ad7d217", uri="sip:192.168.0.3", response="e2144ffda83e96ef1d564496b47d26a1", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

  
<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjEC0VOFXKTCu9tih.uD7abn6xfbcJ5tIM;received=192.168.0.10;rport=5060
From: <sip:0110@192.168.0.3>;tag=aHgaBxKFqtTjzfv6BJWBFE72SRB-TKQo
To: <sip:0110@192.168.0.3>;tag=as6ddcbf4e
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23502 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
C
  
ontact: <sip:0110@192.168.0.10:5060>;expires=300
Date: Thu, 02 Aug 2012 19:00:40 GMT
Content-Length: 0


<------------>

  
Scheduling destruction of SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' in 32000 ms (Method: REGISTER)

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjbcl8SwL0OWZyiQGF49p2gNaXAA3z3.bf
Max-Forwards: 70
From: <sip:0210@192.168.0.3>;tag=HaPhs43L9o1VlsQ3laTP.F1u55SMQM8v
To: <sip:0210@192.168.0.3>
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30505 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0210@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjbcl8SwL0OWZyiQGF49p2gNaXAA3z3.bf;received=192.168.0.10;rport=5060
From: <sip:0210@192.168.0.3>;tag=HaPhs43L9o1VlsQ3laTP.F1u55SMQM8v
To: <sip:0210@192.168.0.3>;tag=as60105dbb
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30505 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="090d3546"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' in 32000 ms (Method: REGISTER)

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPj17ZN5sci6JPhhexDLvZjMdyfzz5gv3-e
Max-Forwards: 70
From: <sip:0210@192.168.0.3>;tag=HaPhs43L9o1VlsQ3laTP.F1u55SMQM8v
To: <sip:0210@192.168.0.3>
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30506 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0210@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0210", realm="asterisk", nonce="090d3546", uri="sip:192.168.0.3", response="43cb1145485a10d60267804c69be521f", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

  
<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPj17ZN5sci6JPhhexDLvZjMdyfzz5gv3-e;received=192.168.0.10;rport=5060
From: <sip:0210@192.168.0.3>;tag=HaPhs43L9o1VlsQ3laTP.F1u55SMQM8v
To: <sip:0210@192.168.0.3>;tag=as60105dbb
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30506 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
C
  
ontact: <sip:0210@192.168.0.10:5060>;expires=300
Date: Thu, 02 Aug 2012 19:00:40 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' in 32000 ms (Method: REGISTER)

  
  == Using SIP RTP CoS mark 5
Audio is at 192.168.0.3 port 12168
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.10:5060:
INVITE sip:0210@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK67528369;rport
Max-Forwards: 70
From: "0411" <sip:0411@192.168.0.3>;tag=as0a81c681
To: <sip:0210@192.168.0.10:5060>
Contact: <sip:0411@192.168.0.3>
Call-ID: 00ac551a6e23ec4e5afa3795050f9d70@192.168.0.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 02 Aug 2012 19:00:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 408684123 408684123 IN IP4 192.168.0.3
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.0.3
t=0 0
m=audio 12168 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

  
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;received=192.168.0.3;branch=z9hG4bK67528369
Call-ID: 00ac551a6e23ec4e5afa3795050f9d70@192.168.0.3
From: "0411" <sip:0411@192.168.0.3>;tag=as0a81c681
To: <sip:0210@192.168.0.10>
CSeq: 102 INVITE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---

  
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;received=192.168.0.3;branch=z9hG4bK67528369
Call-ID: 00ac551a6e23ec4e5afa3795050f9d70@192.168.0.3
From: "0411" <sip:0411@192.168.0.3>;tag=as0a81c681
To: <sip:0210@192.168.0.10>;tag=sbxlnkgezWY-yt.HXqnS1pGmEYpTOSOd
CSeq: 102 INVITE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.10
Transmitting (no NAT) to 192.168.0.10:5060:
ACK sip:0210@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK67528369;rport
Max-Forwards: 70
From: "0411" <sip:0411@192.168.0.3>;tag=as0a81c681
To: <sip:0210@192.168.0.10:5060>;tag=sbxlnkgezWY-yt.HXqnS1pGmEYpTOSOd
Contact: <sip:0411@192.168.0.3>
Call-ID: 00ac551a6e23ec4e5afa3795050f9d70@192.168.0.3
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.6
Content-Length: 0


---

  
  == Using SIP RTP CoS mark 5

  
Really destroying SIP dialog '00ac551a6e23ec4e5afa3795050f9d70@192.168.0.3' Method: INVITE

  
    -- Got SIP response 486 "Busy Here" back from 192.168.1.15
  == Using SIP RTP CoS mark 5

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.13

  
  == Using SIP RTP CoS mark 5

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.22

  


  
Really destroying SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' Method: REGISTER

  
Really destroying SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' Method: REGISTER

  
Really destroying SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' Method: REGISTER



core show channels 
node2*CLI> 
Channel              Location             State   Application(Data)             

  
SIP/0110-00000019    0901@users:2         Up      Konference(0901)              
SIP/0116-00000018    0901@users:2         Up      Konference(0901)              
SIP/0113-00000017    0901@users:2         Up      Konference(0901)              
SIP/0001-00000016    0901@users:2         Up      Konference(0901)              
SIP/0210-00000007    0411@users:2         Up      Konference(0411)              
SIP/0212-00000004    0411@users:2         Up      Konference(0411)              
SIP/0216-00000003    0411@users:2         Up      Konference(0411)              
SIP/0213-00000002    0410@users:2         Up      Konference(0410)              
8 active channels
8 active calls
12 calls processed

REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjRsBItyPmYOFntRiST1O70gn-wA489EKH
Max-Forwards: 70
From: <sip:0010@192.168.0.3>;tag=8XHyaMHaXs3hUjmsgTXepnNhifKPtjDp
To: <sip:0010@192.168.0.3>
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20808 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0010@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjRsBItyPmYOFntRiST1O70gn-wA489EKH;received=192.168.0.10;rport=5060
From: <sip:0010@192.168.0.3>;tag=8XHyaMHaXs3hUjmsgTXepnNhifKPtjDp
To: <sip:0010@192.168.0.3>;tag=as4c2f02a5
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20808 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="764a4c38"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' in 32000 ms (Method: REGISTER)

  
  == Using SIP RTP CoS mark 5

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjLaLIHj-J6dNtgcKsn2uDvjkdyrV0lysd
Max-Forwards: 70
From: <sip:0010@192.168.0.3>;tag=8XHyaMHaXs3hUjmsgTXepnNhifKPtjDp
To: <sip:0010@192.168.0.3>
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20809 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0010@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0010", realm="asterisk", nonce="764a4c38", uri="sip:192.168.0.3", response="16c22af2eafbdde3f5252caf37df1855", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

  
<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjLaLIHj-J6dNtgcKsn2uDvjkdyrV0lysd;received=192.168.0.10;rport=5060
From: <sip:0010@192.168.0.3>;tag=8XHyaMHaXs3hUjmsgTXepnNhifKPtjDp
To: <sip:0010@192.168.0.3>;tag=as4c2f02a5
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20809 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
C
  
ontact: <sip:0010@192.168.0.10:5060>;expires=300
Date: Thu, 02 Aug 2012 19:05:35 GMT
Content-Length: 0


<------------>

  
Scheduling destruction of SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' in 32000 ms (Method: REGISTER)

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjwEWFslkPgIZpB5TOG532ttJeZm3d4vDJ
Max-Forwards: 70
From: <sip:0110@192.168.0.3>;tag=sBJqUgiSmm5U286jI520rtkAVTiDM-Iy
To: <sip:0110@192.168.0.3>
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23503 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0110@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjwEWFslkPgIZpB5TOG532ttJeZm3d4vDJ;received=192.168.0.10;rport=5060
From: <sip:0110@192.168.0.3>;tag=sBJqUgiSmm5U286jI520rtkAVTiDM-Iy
To: <sip:0110@192.168.0.3>;tag=as3126a69c
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23503 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0f9cc0dc"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' in 32000 ms (Method: REGISTER)

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.14

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjlQ7vX9hvvLQBDrszXlzuqRtFI0cFIqkc
Max-Forwards: 70
From: <sip:0110@192.168.0.3>;tag=sBJqUgiSmm5U286jI520rtkAVTiDM-Iy
To: <sip:0110@192.168.0.3>
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23504 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0110@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0110", realm="asterisk", nonce="0f9cc0dc", uri="sip:192.168.0.3", response="f2f7fb59f5fff291f55ead7d7e7b6918", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjlQ7vX9hvvLQBDrszXlzuqRtFI0cFIqkc;received=192.168.0.10;rport=5060
From: <sip:0110@192.168.0.3>;tag=sBJqUgiSmm5U286jI520rtkAVTiDM-Iy
To: <sip:0110@192.168.0.3>;tag=as3126a69c
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23504 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:0110@192.168.0.10:5060>;expires=300
Date: Thu, 02 Aug 2012 19:05:35 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' in 32000 ms (Method: REGISTER)

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjKYcrqPF954WJy0DRGHUfYwDsJ.rpBRQq
Max-Forwards: 70
From: <sip:0210@192.168.0.3>;tag=4aSULDv6Tx-zpbL5FseGQnvz87dg3Yt1
To: <sip:0210@192.168.0.3>
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30507 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0210@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjKYcrqPF954WJy0DRGHUfYwDsJ.rpBRQq;received=192.168.0.10;rport=5060
From: <sip:0210@192.168.0.3>;tag=4aSULDv6Tx-zpbL5FseGQnvz87dg3Yt1
To: <sip:0210@192.168.0.3>;tag=as108ed550
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30507 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a55bd1e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' in 32000 ms (Method: REGISTER)

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjPGShUIqDiUJe0jUiMb7MhFlm4d4x3WcG
Max-Forwards: 70
From: <sip:0210@192.168.0.3>;tag=4aSULDv6Tx-zpbL5FseGQnvz87dg3Yt1
To: <sip:0210@192.168.0.3>
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30508 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0210@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0210", realm="asterisk", nonce="6a55bd1e", uri="sip:192.168.0.3", response="25758c0e6bf0fe5b5352f77d3fd806bb", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

  
<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjPGShUIqDiUJe0jUiMb7MhFlm4d4x3WcG;received=192.168.0.10;rport=5060
From: <sip:0210@192.168.0.3>;tag=4aSULDv6Tx-zpbL5FseGQnvz87dg3Yt1
To: <sip:0210@192.168.0.3>;tag=as108ed550
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30508 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
C
  
ontact: <sip:0210@192.168.0.10:5060>;expires=300
Date: Thu, 02 Aug 2012 19:05:35 GMT
Content-Length: 0


<------------>

  
Scheduling destruction of SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' in 32000 ms (Method: REGISTER)

  
  == Using SIP RTP CoS mark 5

  
Audio is at 192.168.0.3 port 13188
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

  
Reliably Transmitting (no NAT) to 192.168.0.10:5060:
INVITE sip:0210@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK1b22cc50;rport
Max-Forwards: 70
From: "0411" <sip:0411@192.168.0.3>;tag=as4d12abf7
To: <sip:0210@192.168.0.10:5060>
Contact: <sip:0411@192.168.0.3>
Call-ID: 6efe9e80721236cd6abb54d20fc73f28@192.168.0.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 02 Aug 2012 19:05:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1542761767 1542761767 IN IP4 192.168.0.3
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.0.3
t=0 0
m=audio 13188 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

  
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;received=192.168.0.3;branch=z9hG4bK1b22cc50
Call-ID: 6efe9e80721236cd6abb54d20fc73f28@192.168.0.3
From: "0411" <sip:0411@192.168.0.3>;tag=as4d12abf7
To: <sip:0210@192.168.0.10>
CSeq: 102 INVITE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---

  
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;received=192.168.0.3;branch=z9hG4bK1b22cc50
Call-ID: 6efe9e80721236cd6abb54d20fc73f28@192.168.0.3
From: "0411" <sip:0411@192.168.0.3>;tag=as4d12abf7
To: <sip:0210@192.168.0.10>;tag=xut5fnZEa.V9MllISnbPNqfhOqCCdV0t
CSeq: 102 INVITE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.10
Transmitting (no NAT) to 192.168.0.10:5060:
ACK sip:0210@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK1b22cc50;rport
Max-Forwards: 70
From: "0411" <sip:0411@192.168.0.3>;tag=as4d12abf7
To: <sip:0210@192.168.0.10:5060>;tag=xut5fnZEa.V9MllISnbPNqfhOqCCdV0t
Contact: <sip:0411@192.168.0.3>
Call-ID: 6efe9e80721236cd6abb54d20fc73f28@192.168.0.3
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.6
Content-Length: 0


---

  
  == Using SIP RTP CoS mark 5

  
Really destroying SIP dialog '6efe9e80721236cd6abb54d20fc73f28@192.168.0.3' Method: INVITE

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.22

  
  == Using SIP RTP CoS mark 5

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.13

  
  == Using SIP RTP CoS mark 5

  
    -- Got SIP response 486 "Busy Here" back from 192.168.1.15

  

***************************************************************************
after changing system date
***************************************************************************


<------------->

  
Really destroying SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' Method: REGISTER

  
Really destroying SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' Method: REGISTER

  
Really destroying SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' Method: REGISTER

  

REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjH1A8Pibfx4EfoSwuETzrQKZN1mf4EcS7
Max-Forwards: 70
From: <sip:0010@192.168.0.3>;tag=NME7286DG-1qPK48L3F-2OGUsMYNI4or
To: <sip:0010@192.168.0.3>
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20810 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0010@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjH1A8Pibfx4EfoSwuETzrQKZN1mf4EcS7;received=192.168.0.10;rport=5060
From: <sip:0010@192.168.0.3>;tag=NME7286DG-1qPK48L3F-2OGUsMYNI4or
To: <sip:0010@192.168.0.3>;tag=as05a7cad8
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20810 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36f268eb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' in 32000 ms (Method: REGISTER)

  
    -- Registered SIP '0012' at 192.168.0.22 port 5060

  
    -- Registered SIP '0013' at 192.168.0.13 port 5060
    -- Registered SIP '0112' at 192.168.0.22 port 5060
    -- Registered SIP '0113' at 192.168.0.13 port 5060

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjUu6LFrkKi1XkijgtNLkCtnAUPE75A5ug
Max-Forwards: 70
From: <sip:0110@192.168.0.3>;tag=cr59HxVohKv.DMfLqGsZdDN5N7eVvGn6
To: <sip:0110@192.168.0.3>
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23505 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0110@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---

  
Sending to 192.168.0.10 : 5060 (no NAT)

  
<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjUu6LFrkKi1XkijgtNLkCtnAUPE75A5ug;received=192.168.0.10;rport=5060
From: <sip:0110@192.168.0.3>;tag=cr59HxVohKv.DMfLqGsZdDN5N7eVvGn6
To: <sip:0110@192.168.0.3>;tag=as53366985
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23505 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10435a52"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' in 32000 ms (Method: REGISTER)

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjbM-GoxL42PMkKfy4DhKq3bWMuMqAK5Nv
Max-Forwards: 70
From: <sip:0210@192.168.0.3>;tag=ure51D9SvnwkEtO43XlMm8T3svctjyrQ
To: <sip:0210@192.168.0.3>
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30509 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0210@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjbM-GoxL42PMkKfy4DhKq3bWMuMqAK5Nv;received=192.168.0.10;rport=5060
From: <sip:0210@192.168.0.3>;tag=ure51D9SvnwkEtO43XlMm8T3svctjyrQ
To: <sip:0210@192.168.0.3>;tag=as4d820e4f
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30509 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c956712"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' in 32000 ms (Method: REGISTER)

  
    -- Registered SIP '0213' at 192.168.0.13 port 5060

  
  == Using SIP RTP CoS mark 5

  
    -- Registered SIP '0212' at 192.168.0.22 port 5060

  
    -- Registered SIP '0001' at 192.168.0.14 port 5060

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.13

  
    -- Registered SIP '0016' at 192.168.1.15 port 5060

  
    -- Registered SIP '0116' at 192.168.1.15 port 5060

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjj4raa2KAtpXF6jI7If4DgtbAYU.cc6aU
Max-Forwards: 70
From: <sip:0010@192.168.0.3>;tag=NME7286DG-1qPK48L3F-2OGUsMYNI4or
To: <sip:0010@192.168.0.3>
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20811 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0010@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0010", realm="asterisk", nonce="36f268eb", uri="sip:192.168.0.3", response="787e9c8adc823dedfb73234c9334aca7", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

  
    -- Registered SIP '0010' at 192.168.0.10 port 5060

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjj4raa2KAtpXF6jI7If4DgtbAYU.cc6aU;received=192.168.0.10;rport=5060
From: <sip:0010@192.168.0.3>;tag=NME7286DG-1qPK48L3F-2OGUsMYNI4or
To: <sip:0010@192.168.0.3>;tag=as05a7cad8
Call-ID: Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp
CSeq: 20811 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:0010@192.168.0.10:5060>;expires=300
Date: Fri, 03 Aug 2012 19:06:31 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' in 32000 ms (Method: REGISTER)

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjQJqqFVxRPmlnd.9X5i4LewpPI4YCx7Go
Max-Forwards: 70
From: <sip:0110@192.168.0.3>;tag=cr59HxVohKv.DMfLqGsZdDN5N7eVvGn6
To: <sip:0110@192.168.0.3>
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23506 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0110@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0110", realm="asterisk", nonce="10435a52", uri="sip:192.168.0.3", response="fb4b2f286add4c77f7fd42a1bedc8c7a", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)
    -- Registered SIP '0110' at 192.168.0.10 port 5060

<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjQJqqFVxRPmlnd.9X5i4LewpPI4YCx7Go;received=192.168.0.10;rport=5060
From: <sip:0110@192.168.0.3>;tag=cr59HxVohKv.DMfLqGsZdDN5N7eVvGn6
To: <sip:0110@192.168.0.3>;tag=as53366985
Call-ID: D5urcviuBCLh9ohm2HFdruy1v3pg0myN
CSeq: 23506 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:0110@192.168.0.10:5060>;expires=300
Date: Fri, 03 Aug 2012 19:06:31 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' in 32000 ms (Method: REGISTER)

  
  == Using SIP RTP CoS mark 5

  
    -- Registered SIP '0216' at 192.168.1.15 port 5060

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.14

  
  == Using SIP RTP CoS mark 5

  
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;rport;branch=z9hG4bKPjE8uhCPuRAW-qg1EYuuMTyrBIuz.O-FL9
Max-Forwards: 70
From: <sip:0210@192.168.0.3>;tag=ure51D9SvnwkEtO43XlMm8T3svctjyrQ
To: <sip:0210@192.168.0.3>
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30510 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: <sip:0210@192.168.0.10:5060>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0210", realm="asterisk", nonce="7c956712", uri="sip:192.168.0.3", response="4289e9fb60f3e43e34300ccedeebefa0", algorithm=MD5
Content-Length:  0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.10 : 5060 (no NAT)

  
    -- Registered SIP '0210' at 192.168.0.10 port 5060

  
<--- Transmitting (no NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bKPjE8uhCPuRAW-qg1EYuuMTyrBIuz.O-FL9;received=192.168.0.10;rport=5060
From: <sip:0210@192.168.0.3>;tag=ure51D9SvnwkEtO43XlMm8T3svctjyrQ
To: <sip:0210@192.168.0.3>;tag=as4d820e4f
Call-ID: ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz
CSeq: 30510 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
C
  
ontact: <sip:0210@192.168.0.10:5060>;expires=300
Date: Fri, 03 Aug 2012 19:06:31 GMT
Content-Length: 0


<------------>

  
Scheduling destruction of SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' in 32000 ms (Method: REGISTER)

  
    -- Got SIP response 486 "Busy Here" back from 192.168.0.22

  
  == Using SIP RTP CoS mark 5

  
    -- Got SIP response 486 "Busy Here" back from 192.168.1.15

  
  == Using SIP RTP CoS mark 5
Audio is at 192.168.0.3 port 17706
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.10:5060:
INVITE sip:0210@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK11becbb1;rport
Max-Forwards: 70
From: "0411" <sip:0411@192.168.0.3>;tag=as515c5383
To: <sip:0210@192.168.0.10:5060>
Contact: <sip:0411@192.168.0.3>
Call-ID: 5885ccbd772fbd866fa8f8ce5095efd9@192.168.0.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.6
Date: Fri, 03 Aug 2012 19:06:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 218483247 218483247 IN IP4 192.168.0.3
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.0.3
t=0 0
m=audio 17706 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

  
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;received=192.168.0.3;branch=z9hG4bK11becbb1
Call-ID: 5885ccbd772fbd866fa8f8ce5095efd9@192.168.0.3
From: "0411" <sip:0411@192.168.0.3>;tag=as515c5383
To: <sip:0210@192.168.0.10>
CSeq: 102 INVITE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---

  
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;received=192.168.0.3;branch=z9hG4bK11becbb1
Call-ID: 5885ccbd772fbd866fa8f8ce5095efd9@192.168.0.3
From: "0411" <sip:0411@192.168.0.3>;tag=as515c5383
To: <sip:0210@192.168.0.10>;tag=VNm5JqjcuZcAxuHPCB6iD3v.Fw6UidNo
CSeq: 102 INVITE
Content-Length:  0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 192.168.0.10
Transmitting (no NAT) to 192.168.0.10:5060:
ACK sip:0210@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK11becbb1;rport
Max-Forwards: 70
From: "0411" <sip:0411@192.168.0.3>;tag=as515c5383
To: <sip:0210@192.168.0.10:5060>;tag=VNm5JqjcuZcAxuHPCB6iD3v.Fw6UidNo
Contact: <sip:0411@192.168.0.3>
Call-ID: 5885ccbd772fbd866fa8f8ce5095efd9@192.168.0.3
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.6
Content-Length: 0


---

  
Really destroying SIP dialog '5885ccbd772fbd866fa8f8ce5095efd9@192.168.0.3' Method: INVITE

  

  core  show channel SIP/0001-00000016
node2*CLI> 
 -- General --
           Name: SIP/0001-00000016
           Type: SIP
       UniqueID: 1343933224.23
      Caller ID: 0001
 Caller ID Name: (N/A)
    DNID Digits: 0901
       Language: en
          State: Up (6)
          Rings: 0
  NativeFormats: 0x1000 (g722)
    WriteFormat: 0x1000 (g722)
     ReadFormat: 0x1000 (g722)
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: 28
      Frames in: 59407
     Frames out: 81313
 Time to Hangup: 0
   Elapsed Time: 24h19m45s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: users
      Extension: 0901
       Priority: 2
     Call Group: 0
   Pickup Group: 0
    Application: Konference
           Data: 0901
    Blocking in: ast_waitfor_nandfds
      Variables:
SIPCALLID=tw-DEhUyx-Vude5GEf-fynbATF3pQ25D
SIPDOMAIN=192.168.0.3
SIPURI=sip:0001@192.168.0.14:5060

  CDR Variables:
level 1: dnid=0901
level 1: clid=0001
level 1: src=0001
level 1: dst=0901
level 1: dcontext=users
level 1: channel=SIP/0001-00000016
level 1: lastapp=Konference
level 1: lastdata=0901
level 1: start=2012-08-03 00:17:04
level 1: answer=2012-08-03 00:17:04
level 1: duration=87584
level 1: billsec=87584
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1343933224.23

Really destroying SIP dialog 'Kc12CoeFxSKAH4-Q5WlIxLNwvnyJkMHp' Method: REGISTER

  
Really destroying SIP dialog 'D5urcviuBCLh9ohm2HFdruy1v3pg0myN' Method: REGISTER

   
Really destroying SIP dialog 'ugU-P1fqqx-ek9s.k5z1qIXx2RAW50Iz' Method: REGISTER

  core  show channel  SIP/0110-00000019
node2*CLI> 
 -- General --
           Name: SIP/0110-00000019
           Type: SIP
       UniqueID: 1343933224.26
      Caller ID: 0901
 Caller ID Name: (N/A)
    DNID Digits: (N/A)
       Language: en
          State: Up (6)
          Rings: 0
  NativeFormats: 0x1000 (g722)
    WriteFormat: 0x1000 (g722)
     ReadFormat: 0x1000 (g722)
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: 46
      Frames in: 55339
     Frames out: 106814
 Time to Hangup: 0
   Elapsed Time: 24h20m5s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: users
      Extension: 0901
       Priority: 2
     Call Group: 0
   Pickup Group: 0
    Application: Konference
           Data: 0901
    Blocking in: ast_waitfor_nandfds
      Variables:
host=dynamic
type=friend
quality=no
secret=0110
SIPCALLID=5725813e69ce4ae26da4e6ad76eade69@192.168.0.3

  CDR Variables:
level 1: clid=0901
level 1: src=0901
level 1: dst=0901
level 1: dcontext=users
level 1: channel=SIP/0110-00000019
level 1: lastapp=Konference
level 1: lastdata=0901
level 1: start=2012-08-03 00:17:04
level 1: duration=87604
level 1: billsec=0
level 1: disposition=NO ANSWER
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1343933224.26


  core  show channel SIP/0001-00000016 
node2*CLI> 

  
  * SIP Call

  
  Curr. trans. direction:  Incoming

  
  Call-ID:                tw-DEhUyx-Vude5GEf-fynbATF3pQ25D

  
  Owner channel ID:       SIP/0001-00000016

  
  Our Codec Capability:   4096

  
  Non-Codec Capability (DTMF):   1

  
  Their Codec Capability:   4110

  
  Joint Codec Capability:   4096

  
  Format:                 0x1000 (g722)

  
  T.38 support            No

  
  Video support           No

  
  MaxCallBR:              384 kbps

  
  Theoretical Address:    192.168.0.14:5060

  
  Received Address:       192.168.0.14:5060

  
  SIP Transfer mode:      open

  
  NAT Support:            RFC3581

  
  Audio IP:               192.168.0.3 (local)

  
  Our Tag:                as65db3dee

  
  Their Tag:              TnngYjrETLOyQYXThpt.CnlFC4MxIhkm

  
  SIP User agent:         PJSUA v1.8/arm-none-linux-gnueabi

  
  Username:               0001

  
  Peername:               0001

  
  Original uri:           sip:0001@192.168.0.14:5060

  
  Caller-ID:              0001

  
  Need Destroy:           No

  
  Last Message:           Rx: ACK

  
  Promiscuous Redir:      No

  
  Route:                  sip:0001@192.168.0.14:5060

  
  DTMF Mode:              rfc2833

  
  SIP Options:            100rel norefersub replaces replace timer 

  
  Session-Timer:          Inactive

  


  sip  show channel tw-DEhUyx-Vude5GEf-fynbATF3pQ25D 
node2*CLI> 

  
  * SIP Call

  
  Curr. trans. direction:  Outgoing

  
  Call-ID:                5725813e69ce4ae26da4e6ad76eade69@192.168.0.3

  
  Owner channel ID:       SIP/0110-00000019

  
  Our Codec Capability:   4096

  
  Non-Codec Capability (DTMF):   1

  
  Their Codec Capability:   4096

  
  Joint Codec Capability:   4096

  
  Format:                 0x1000 (g722)

  
  T.38 support            No

  
  Video support           No

  
  MaxCallBR:              384 kbps

  
  Theoretical Address:    192.168.0.10:5060

  
  Received Address:       192.168.0.10:5060
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               192.168.0.3 (local)
  Our Tag:                as685ebb31
  Their Tag:              8DEMCabPHdA1TWmYQBKPIV8awP1lDic9

  
  SIP User agent:         
  Username:               0110
  Peername:               0110
  Original uri:           sip:0110@192.168.0.10:5060

  
  Need Destroy:           No
  Last Message:           Tx: ACK

  
  Promiscuous Redir:      No
  Route:                  sip:0110@192.168.0.10:5060
  DTMF Mode:              rfc2833

  
  SIP Options:            (none)
  Session-Timer:          Inactive
 
  sip show peer 0001
node2*CLI> 

  * Name       : 0001
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : users
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 232
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  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       : 
  Addr->IP     : 192.168.0.14 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 0001
  SIP Options  : (none)
  Codecs       : 0x1000 (g722)
  Codec Order  : (g722:20)
  Auto-Framing :  No 
  100 on REG   : No
  Status       : Unmonitored
  Useragent    : PJSUA v1.8/arm-none-linux-gnueabi
  Reg. Contact : sip:0001@192.168.0.14:5060
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uac
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  Parkinglot   : 

  sip show peer 0001  
node2*CLI> 
  * Name       : 0110
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : users
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 227
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  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       : 
  Addr->IP     : 192.168.0.10 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 0110
  SIP Options  : (none)
  Codecs       : 0x1000 (g722)
  Codec Order  : (g722:20)
  Auto-Framing :  No 
  100 on REG   : No
  Status       : Unmonitored
  Useragent    : PJSUA v1.8/arm-none-linux-gnueabi
  Reg. Contact : sip:0110@192.168.0.10:5060
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  Parkinglot   : 
  
<--- SIP read from UDP:192.168.0.10:5060 --->
<------------->
<--- SIP read from UDP:192.168.0.10:5060 --->
<------------->
core show channels 
node2*CLI> 
Channel              Location             State   Application(Data)             
SIP/0110-00000019    0901@users:2         Up      Konference(0901)              
SIP/0116-00000018    0901@users:2         Up      Konference(0901)              
SIP/0113-00000017    0901@users:2         Up      Konference(0901)              
SIP/0001-00000016    0901@users:2         Up      Konference(0901)              
SIP/0210-00000007    0411@users:2         Up      Konference(0411)              
SIP/0212-00000004    0411@users:2         Up      Konference(0411)              
SIP/0216-00000003    0411@users:2         Up      Konference(0411)              
SIP/0213-00000002    0410@users:2         Up      Konference(0410)              
8 active channels
8 active calls
12 calls processed

To start with does this issue lie with Asterisk or the sip usragent …?
If the problem is with asterisk & or sip configuration is there a resoultion…?

thanks in advance
Pranav

2

is there no one else facing this issue?
Any help will be appreciated

Thanks & Regards
Pranav Thipse

3

It is rather important whether this is elapsed time or time of day related. Asterisk has nothing time of day related for established calls. It can have elapsed time related issues, although anything explicit would normally be less than an hour, and would be associated with timeout messages, or SIP exchanges.

The log has a comment about changing the system date. That is a very disruptive thing to do to a real time system. Can you confirm you did it to try to get the log quickly, and that the fault does not depend on it.

The log is very noisy; there are a lot of rejected calls (486 Busy), which are, presumably, unrelated to the problem. It would help if you could eliminate unrelated calls, possibly eliminate registers for everything except the phone that experiences the problem, and try to cut it down to just the start of the call and the time around when the fault presents.

With such a long dely to failure, I would look for:

  • accumulation of timing skew;
  • internl firewall, or network switch, forgetting the “connection”;
  • something re-assigning a dynamic address.