Call drop after ringing when receiver pick up

Two telephone are registered successfully in asterisk.sip show peers shows it is specified to an Ip. when call from one to another , ringing is successfully done,but when try to answer the call by picking up the receiver call drops.

sip.conf and extensions.conf files are as follows

#####sip.conf #####
[general]
context=default
bindaddr=0.0.0.0
bindport=5060
disallow=all
allow=alaw
allow=g729
srvlookup=yes
auth_message_requests=no
accept_outofcall_message=yes
;outofcall_message_context=dpma_message_context
outofcall_message_context=internal
auth_message_requests=no
callcounter=yes
realm=asterisk.rng.com
domain=172.16.200.35

[1310]
type=friend
host=dynamic
;host=172.16.200.20
context=internal
username=1310
secret=1310
mailbox=1234@default
nat=no
qualify=yes

[1311]
type=friend
host=dynamic
;host=172.16.200.20
context=internal
username=1311
secret=1311
mailbox=1234@default
nat=no
qualify=yes

[authentication]
auth=1310:1310@172.16.200.35
auth=1311:1311@172.16.200.35

######extensions.conf#######
[incoming]
; Ring on extension 1310 and 1311.
exten => s,1,Answer()
exten => s,n,Dial(SIP/1310,150,r,t,)

; Still not answered? Pass unanswered calls to voicemail
exten => s,n,Voicemail(100,u)
exten => s,n,Hangup

[outgoing]
; Outbound calls can be routed based on the number of digits dialled (or the value of the first few digits)
exten => _XXXX,1,Dial(SIP/${EXTEN})

[internal]
; Calls between employees (between extensions)
exten => _XXXX,1,Dial(SIP/${EXTEN})

; Calls to ext 1310
exten => 1310,1,Dial(SIP/1310,20)
exten => 1310,n,VoiceMail(1310,u)
exten => 1310,n,Hangup

; Calls to ext 1311
exten => 1311,1,Dial(SIP/1311,20)
exten => 1311,n,Hangup

[phones]
include => internal

Logs?

asterisk*CLI> sip show peers

[Name/username             Host                                    Dyn Forcerport ACL Port     Status      Description                      
1310/1310                 172.16.201.20                            D                 5060     OK (63 ms)                                   
1311/1311                 172.16.201.20                            D                 5060     OK (27 ms)                                   

 
 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

asterisk*CLI> sip show peer 1310

  * Name       : 1310

   Description  : 
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>

   Context      : internal
  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      : 1234@default
  VM Extension : asterisk

 
   LastMsgsSent : 0/0
  Call limit   : 2147483647
  Max forwards : 0

 
   Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps

 
   Expire       : 3561
  Insecure     : no
  Force rport  : No
  Symmetric RTP: No

 
   ACL          : No
  DirectMedACL : 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 : Yes

 
   DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 172.16.200.20

 
   Addr->IP     : 172.16.201.20:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP

 
   Def. Username: 1310
  SIP Options  : (none)
  Codecs       : (alaw|g729)
  Codec Order  : (alaw:20,g729:20)

 
   Auto-Framing :  No 
  Status       : OK (63 ms)

 
   Useragent    : Asterisk
  Reg. Contact : sip:1310@172.16.201.20;transport=udp
  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


asterisk*CLI> show peer 1311
  * Name       : 1311

   Description  : 
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>

   Context      : internal
  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      : 1234@default
 
   VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 2147483647
 
   Max forwards : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps

   Expire       : 3556
  Insecure     : no
  Force rport  : No
  Symmetric RTP: No
 
   ACL          : No
  DirectMedACL : 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 : Yes
 
   DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 172.16.200.20

   Addr->IP     : 172.16.201.20:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP

   Def. Username: 1311
  SIP Options  : (none)
  Codecs       : (alaw|g729)
  Codec Order  : (alaw:20,g729:20)

   Auto-Framing :  No 
  Status       : OK (27 ms)
  Useragent    : Asterisk

   Reg. Contact : sip:1311@172.16.201.20;transport=udp
  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


 
<--- SIP read from UDP:172.16.201.20:5060 --->
INVITE sip:1310@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bK3833bd693beb9b548
Max-Forwards: 70
From: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
To: 1310 <sip:1310@172.16.200.35:5060>
Call-ID: 4077a06730bb3452
CSeq: 1507304249 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Contact: 1311 <sip:1311@172.16.201.20;transport=udp>
Supported: timer, replaces
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length: 673

v=0
o=MxSIP 0 0 IN IP4 172.16.201.20
s=SIP Call
c=IN IP4 172.16.201.20
t=0 0
m=audio 53462 RTP/AVP 9 8 18 0 98 97 2 96 4 15 100 106 107 114 119 101
a=rtpmap:9 G722/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:4 G723/8000
a=rtpmap:15 G728/8000
a=rtpmap:100 G729E/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:114 iLBC/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=yes
a=fmtp:114 mode=20
a=fmtp:101 0-15,144,149,159
a=ptime:20
a=sendrecv
a=silenceSupp:on - - - -
<------------->

 
 --- (13 headers 28 lines) ---

 
 Sending to 172.16.201.20:5060 (no NAT)
Sending to 172.16.201.20:5060 (no NAT)

 
 Using INVITE request as basis request - 4077a06730bb3452

 
 Found peer '1311' for '1311' from 172.16.201.20:5060

 
 
<--- Reliably Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bK3833bd693beb9b548;received=172.16.201.20
From: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
To: 1310 <sip:1310@172.16.200.35:5060>;tag=as6b22e7c3
Call-ID: 4077a06730bb3452
CSeq: 1507304249 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.rng.com", nonce="05243578"
Content-Length: 0




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

 
 Scheduling destruction of SIP dialog '4077a06730bb3452' in 6400 ms (Method: INVITE)

 
 
<--- SIP read from UDP:172.16.201.20:5060 --->
ACK sip:1310@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bK3833bd693beb9b548
Max-Forwards: 70
From: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
To: 1310 <sip:1310@172.16.200.35:5060>;tag=as6b22e7c3
Call-ID: 4077a06730bb3452
CSeq: 1507304249 ACK
User-Agent: Asterisk
Content-Length: 0

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

 
 --- (9 headers 0 lines) ---

 
 
<--- SIP read from UDP:172.16.201.20:5060 --->
INVITE sip:1310@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKff751513788542156
Max-Forwards: 70
From: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
To: 1310 <sip:1310@172.16.200.35:5060>
Call-ID: 4077a06730bb3452
CSeq: 1507304250 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Authorization: Digest username="1311",realm="asterisk.rng.com",nonce="05243578",uri="sip:1310@172.16.200.35:5060",response="3f41c6de8545b70720b7ba7eb397ef0c",algorithm=MD5
Contact: 1311 <sip:1311@172.16.201.20;transport=udp>
Supported: timer, replaces
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length: 673

v=0
o=MxSIP 0 0 IN IP4 172.16.201.20
s=SIP Call
c=IN IP4 172.16.201.20
t=0 0
m=audio 53462 RTP/AVP 9 8 18 0 98 97 2 96 4 15 100 106 107 114 119 101
a=rtpmap:9 G722/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:4 G723/8000
a=rtpmap:15 G728/8000
a=rtpmap:100 G729E/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:114 iLBC/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=yes
a=fmtp:114 mode=20
a=fmtp:101 0-15,144,149,159
a=ptime:20
a=sendrecv
a=silenceSupp:on - - - -
<------------->

 
 --- (14 headers 28 lines) ---

 
 Sending to 172.16.201.20:5060 (no NAT)
 Using INVITE request as basis request - 4077a06730bb3452
 Found peer '1311' for '1311' from 172.16.201.20:5060
   == Using SIP RTP CoS mark 5
 Found RTP audio format 9
 Found RTP audio format 8
 Found RTP audio format 18
 Found RTP audio format 0
 Found RTP audio format 98
 Found RTP audio format 97
 Found RTP audio format 2
 Found RTP audio format 96
 Found RTP audio format 4
 Found RTP audio format 15
 Found RTP audio format 100
 Found RTP audio format 106
 Found RTP audio format 107
 Found RTP audio format 114
 Found RTP audio format 119
 Found RTP audio format 101
 Found unknown media description format G722 for ID 9
 Found audio description format PCMA for ID 8
 Found audio description format G729 for ID 18
 Found audio description format PCMU for ID 0
 Found unknown media description format G726-16 for ID 98
 Found unknown media description format G726-24 for ID 97
 Found audio description format G726-32 for ID 2
 Found unknown media description format G726-40 for ID 96
 Found audio description format G723 for ID 4
 Found unknown media description format G728 for ID 15
 Found unknown media description format G729E for ID 100
 Found unknown media description format BV16 for ID 106
 Found unknown media description format BV32 for ID 107
 Found audio description format iLBC for ID 114
 Found unknown media description format AMR-WB for ID 119
 Found audio description format telephone-event for ID 101

 Capabilities: us - (alaw|g729), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (alaw|g729)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

 Peer audio RTP is at port 172.16.201.20:53462
 Looking for 1310 in internal (domain 172.16.200.35)
 list_route: hop: <sip:1311@172.16.201.20;transport=udp>

  
<--- Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKff751513788542156;received=172.16.201.20
From: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
To: 1310 <sip:1310@172.16.200.35:5060>
Call-ID: 4077a06730bb3452
CSeq: 1507304250 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1310@172.16.200.35:5060>
Content-Length: 0




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

 
 
 
     -- Executing [1310@internal:1] e[1;36mDiale[0m("e[1;35mSIP/1311-00000068e[0m", "e[1;35mSIP/1310,20e[0m") in new stack

 
   == Using SIP RTP CoS mark 5
 Audio is at 21680
 Adding codec 100004 (alaw) to SDP
 Adding codec 100008 (g729) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 
 Reliably Transmitting (no NAT) to 172.16.201.20:5060:
INVITE sip:1310@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK7806b143
Max-Forwards: 70
From: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
To: <sip:1310@172.16.201.20;transport=udp>
Contact: <sip:1311@172.16.200.35:5060>
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 31 Dec 2014 01:27:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284


v=0
o=root 2122600447 2122600447 IN IP4 172.16.200.35
s=Asterisk PBX 11.7.0
c=IN IP4 172.16.200.35
t=0 0
m=audio 21680 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


---

 
     -- Called SIP/1310

 
 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK7806b143
From: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
To: <sip:1310@172.16.201.20;transport=udp>;tag=216833047
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Contact: 1310 <sip:1310@172.16.201.20;transport=udp>
Content-Length: 0

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

 
 --- (9 headers 0 lines) ---

 
 list_route: hop: <sip:1310@172.16.201.20;transport=udp>

 
     -- SIP/1310-00000069 is ringing

 
 
<--- Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKff751513788542156;received=172.16.201.20
From: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
To: 1310 <sip:1310@172.16.200.35:5060>;tag=as5244ed4a
Call-ID: 4077a06730bb3452
CSeq: 1507304250 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1310@172.16.200.35:5060>
Content-Length: 0



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

 
 Reliably Transmitting (no NAT) to 172.16.201.20:5060:
OPTIONS sip:1310@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK2a50603f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.200.35>;tag=as1588b9b0
To: <sip:1310@172.16.201.20;transport=udp>
Contact: <sip:asterisk@172.16.200.35:5060>
Call-ID: 63bc0c2b21c7be994de9a85d04284c60@172.16.200.35:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 31 Dec 2014 01:27:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK2a50603f
From: "asterisk" <sip:asterisk@172.16.200.35>;tag=as1588b9b0
To: <sip:1310@172.16.201.20;transport=udp>;tag=3931411529
Call-ID: 63bc0c2b21c7be994de9a85d04284c60@172.16.200.35:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Content-Length: 0

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

 
 --- (8 headers 0 lines) ---

 
 Really destroying SIP dialog '63bc0c2b21c7be994de9a85d04284c60@172.16.200.35:5060' Method: OPTIONS

 
 Reliably Transmitting (no NAT) to 172.16.201.20:5060:
OPTIONS sip:1311@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK1e95773d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.200.35>;tag=as322695a7
To: <sip:1311@172.16.201.20;transport=udp>
Contact: <sip:asterisk@172.16.200.35:5060>
Call-ID: 70bdba1409d0c92a67ab6bdb09d696cb@172.16.200.35:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 31 Dec 2014 01:27:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

 
 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK1e95773d
From: "asterisk" <sip:asterisk@172.16.200.35>;tag=as322695a7
To: <sip:1311@172.16.201.20;transport=udp>;tag=1786688364
Call-ID: 70bdba1409d0c92a67ab6bdb09d696cb@172.16.200.35:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Content-Length: 0

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

 
 --- (8 headers 0 lines) ---

 
 Really destroying SIP dialog '70bdba1409d0c92a67ab6bdb09d696cb@172.16.200.35:5060' Method: OPTIONS

 
 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK7806b143
From: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
To: <sip:1310@172.16.201.20;transport=udp>;tag=216833047
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Contact: 1310 <sip:1310@172.16.201.20;transport=udp>
Supported: timer, replaces
Content-Type: application/sdp
Content-Length: 295

v=0
o=MxSIP 0 0 IN IP4 172.16.201.20
s=SIP Call
c=IN IP4 172.16.201.20
t=0 0
m=audio 53456 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=yes
a=fmtp:101 0-15,144,149,159
a=ptime:20
a=sendrecv
a=silenceSupp:on - - - -
<------------->

 
 --- (11 headers 14 lines) ---
 Found RTP audio format 8
 Found RTP audio format 18
 Found RTP audio format 101
 Found audio description format PCMA for ID 8
 Found audio description format G729 for ID 18
 Found audio description format telephone-event for ID 101

 Capabilities: us - (alaw|g729), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

 Peer audio RTP is at port 172.16.201.20:53456

 list_route: hop: <sip:1310@172.16.201.20;transport=udp>
 set_destination: Parsing <sip:1310@172.16.201.20;transport=udp> for address/port to send to
 set_destination: set destination to 172.16.201.20:5060

 
 Transmitting (no NAT) to 172.16.201.20:5060:
ACK sip:1310@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK6a37baac
Max-Forwards: 70
From: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
To: <sip:1310@172.16.201.20;transport=udp>;tag=216833047
Contact: <sip:1311@172.16.200.35:5060>
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0

---

 
     -- SIP/1310-00000069 answered SIP/1311-00000068

 
 
 
 Audio is at 29020
 Adding codec 100004 (alaw) to SDP
 Adding codec 100008 (g729) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 
<--- Reliably Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKff751513788542156;received=172.16.201.20
From: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
To: 1310 <sip:1310@172.16.200.35:5060>;tag=as5244ed4a
Call-ID: 4077a06730bb3452
CSeq: 1507304250 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1310@172.16.200.35:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 700912351 700912351 IN IP4 172.16.200.35
s=Asterisk PBX 11.7.0
c=IN IP4 172.16.200.35
t=0 0
m=audio 29020 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

 
     -- Remotely bridging SIP/1311-00000068 and SIP/1310-00000069

 set_destination: Parsing <sip:1310@172.16.201.20;transport=udp> for address/port to send to
 set_destination: set destination to 172.16.201.20:5060

 Audio is at 21680
 Adding codec 100004 (alaw) to SDP
 Adding codec 100008 (g729) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 Reliably Transmitting (no NAT) to 172.16.201.20:5060:
INVITE sip:1310@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK2ef8e743
Max-Forwards: 70
From: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
To: <sip:1310@172.16.201.20;transport=udp>;tag=216833047
Contact: <sip:1311@172.16.200.35:5060>
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 2122600447 2122600448 IN IP4 172.16.201.20
s=Asterisk PBX 11.7.0
c=IN IP4 172.16.201.20
t=0 0
m=audio 53462 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

 
<--- SIP read from UDP:172.16.201.20:5060 --->
BYE sip:1311@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bK89ff0fcdcd5d77c1c
Max-Forwards: 70
From: <sip:1310@172.16.201.20;transport=udp>;tag=216833047
To: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 516833267 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Supported: timer
User-Agent: Asterisk
Content-Length: 0

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

 
 --- (11 headers 0 lines) ---

 Sending to 172.16.201.20:5060 (no NAT)
 Scheduling destruction of SIP dialog '6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bK89ff0fcdcd5d77c1c;received=172.16.201.20
From: <sip:1310@172.16.201.20;transport=udp>;tag=216833047
To: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 516833267 BYE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

   == Spawn extension (internal, 1310, 1) exited non-zero on 'SIP/1311-00000068'
 Scheduling destruction of SIP dialog '4077a06730bb3452' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK2ef8e743
From: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
To: <sip:1310@172.16.201.20;transport=udp>;tag=216833047
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Content-Length: 0

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

 --- (8 headers 0 lines) ---

 set_destination: Parsing <sip:1310@172.16.201.20;transport=udp> for address/port to send to
 set_destination: set destination to 172.16.201.20:5060

 Transmitting (no NAT) to 172.16.201.20:5060:
ACK sip:1310@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK2ef8e743
Max-Forwards: 70
From: <sip:1310@172.16.201.20;transport=udp>;tag=216833047
To: "1311" <sip:1311@172.16.200.35>;tag=as34d4000c
Contact: <sip:1311@172.16.200.35:5060>
Call-ID: 6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0

---

<--- SIP read from UDP:172.16.201.20:5060 --->
ACK sip:1310@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bK5a796023f866e0e64
Max-Forwards: 70
From: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
To: 1310 <sip:1310@172.16.200.35:5060>;tag=as5244ed4a
Call-ID: 4077a06730bb3452
CSeq: 1507304250 ACK
Authorization: Digest username="1311",realm="asterisk.rng.com",nonce="05243578",uri="sip:1310@172.16.200.35:5060",response="3f41c6de8545b70720b7ba7eb397ef0c",algorithm=MD5
User-Agent: Asterisk
Content-Length: 0

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

 
 --- (10 headers 0 lines) ---
 
 set_destination: Parsing <sip:1311@172.16.201.20;transport=udp> for address/port to send to
 set_destination: set destination to 172.16.201.20:5060

 Reliably Transmitting (no NAT) to 172.16.201.20:5060:
BYE sip:1311@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK3e3597bc
Max-Forwards: 70
From: 1310 <sip:1310@172.16.200.35:5060>;tag=as5244ed4a
To: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
Call-ID: 4077a06730bb3452
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.7.0
Proxy-Authorization: Digest username="1311", realm="asterisk.rng.com", algorithm=MD5, uri="sip:172.16.200.35", nonce="05243578", response="8b8b26f72f55f6fb64430a1485a71cb9"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

 Scheduling destruction of SIP dialog '4077a06730bb3452' in 6400 ms (Method: ACK)
 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK3e3597bc
From: 1310 <sip:1310@172.16.200.35:5060>;tag=as5244ed4a
To: 1311 <sip:1311@172.16.200.35>;tag=41d12b6e7f
Call-ID: 4077a06730bb3452
CSeq: 102 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Content-Length: 0

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

 --- (8 headers 0 lines) ---
 
 SIP Response message for INCOMING dialog BYE arrived
 Really destroying SIP dialog '4077a06730bb3452' Method: ACK
 Really destroying SIP dialog '6230d7c0721b4c3a76f61ae2064ae591@172.16.200.35:5060' Method: BYE

<--- SIP read from UDP:172.16.201.20:5060 --->
INVITE sip:1311@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKbc48c5bb209660c8c
Max-Forwards: 70
From: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
To: 1311 <sip:1311@172.16.200.35:5060>
Call-ID: 57e938dda8136fac
CSeq: 382343861 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Contact: 1310 <sip:1310@172.16.201.20;transport=udp>
Supported: timer, replaces
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length: 673

v=0
o=MxSIP 0 0 IN IP4 172.16.201.20
s=SIP Call
c=IN IP4 172.16.201.20
t=0 0
m=audio 53458 RTP/AVP 9 8 18 0 98 97 2 96 4 15 100 106 107 114 119 101
a=rtpmap:9 G722/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:4 G723/8000
a=rtpmap:15 G728/8000
a=rtpmap:100 G729E/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:114 iLBC/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=yes
a=fmtp:114 mode=20
a=fmtp:101 0-15,144,149,159
a=ptime:20
a=sendrecv
a=silenceSupp:on - - - -
<------------->

 
 --- (13 headers 28 lines) ---
 
 Sending to 172.16.201.20:5060 (no NAT)
 Sending to 172.16.201.20:5060 (no NAT)

 Using INVITE request as basis request - 57e938dda8136fac

 Found peer '1310' for '1310' from 172.16.201.20:5060

 
<--- Reliably Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKbc48c5bb209660c8c;received=172.16.201.20
From: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
To: 1311 <sip:1311@172.16.200.35:5060>;tag=as7783e0a7
Call-ID: 57e938dda8136fac
CSeq: 382343861 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.rng.com", nonce="0d0916e3"
Content-Length: 0

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

 Scheduling destruction of SIP dialog '57e938dda8136fac' in 6400 ms (Method: INVITE)
 
<--- SIP read from UDP:172.16.201.20:5060 --->
ACK sip:1311@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKbc48c5bb209660c8c
Max-Forwards: 70
From: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
To: 1311 <sip:1311@172.16.200.35:5060>;tag=as7783e0a7
Call-ID: 57e938dda8136fac
CSeq: 382343861 ACK
User-Agent: Asterisk
Content-Length: 0

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

 --- (9 headers 0 lines) ---

<--- SIP read from UDP:172.16.201.20:5060 --->
INVITE sip:1311@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKed92214e821c72335
Max-Forwards: 70
From: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
To: 1311 <sip:1311@172.16.200.35:5060>
Call-ID: 57e938dda8136fac
CSeq: 382343862 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Authorization: Digest username="1310",realm="asterisk.rng.com",nonce="0d0916e3",uri="sip:1311@172.16.200.35:5060",response="f53a3f7c991170b607f702d6319bd9db",algorithm=MD5
Contact: 1310 <sip:1310@172.16.201.20;transport=udp>
Supported: timer, replaces
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length: 673

v=0
o=MxSIP 0 0 IN IP4 172.16.201.20
s=SIP Call
c=IN IP4 172.16.201.20
t=0 0
m=audio 53458 RTP/AVP 9 8 18 0 98 97 2 96 4 15 100 106 107 114 119 101
a=rtpmap:9 G722/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:4 G723/8000
a=rtpmap:15 G728/8000
a=rtpmap:100 G729E/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:114 iLBC/8000
a=rtpmap:119 AMR-WB/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=yes
a=fmtp:114 mode=20
a=fmtp:101 0-15,144,149,159
a=ptime:20
a=sendrecv
a=silenceSupp:on - - - -
<------------->

 
 --- (14 headers 28 lines) ---

 Sending to 172.16.201.20:5060 (no NAT)
 
 Using INVITE request as basis request - 57e938dda8136fac
 Found peer '1310' for '1310' from 172.16.201.20:5060

   == Using SIP RTP CoS mark 5

 Found RTP audio format 9
 Found RTP audio format 8
 Found RTP audio format 18
 Found RTP audio format 0
 Found RTP audio format 98
 Found RTP audio format 97
 Found RTP audio format 2
 Found RTP audio format 96
 Found RTP audio format 4
 Found RTP audio format 15
 Found RTP audio format 100
 Found RTP audio format 106
 Found RTP audio format 107
 Found RTP audio format 114
 Found RTP audio format 119
 Found RTP audio format 101
 Found unknown media description format G722 for ID 9
 Found audio description format PCMA for ID 8
 Found audio description format G729 for ID 18
 Found audio description format PCMU for ID 0
 Found unknown media description format G726-16 for ID 98
 Found unknown media description format G726-24 for ID 97
 Found audio description format G726-32 for ID 2
 Found unknown media description format G726-40 for ID 96
 Found audio description format G723 for ID 4
 Found unknown media description format G728 for ID 15
 Found unknown media description format G729E for ID 100
 Found unknown media description format BV16 for ID 106
 Found unknown media description format BV32 for ID 107
 Found audio description format iLBC for ID 114
 Found unknown media description format AMR-WB for ID 119
 Found audio description format telephone-event for ID 101

 Capabilities: us - (alaw|g729), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (alaw|g729)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

 Peer audio RTP is at port 172.16.201.20:53458
 Looking for 1311 in internal (domain 172.16.200.35)

 list_route: hop: <sip:1310@172.16.201.20;transport=udp>

<--- Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKed92214e821c72335;received=172.16.201.20
From: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
To: 1311 <sip:1311@172.16.200.35:5060>
Call-ID: 57e938dda8136fac
CSeq: 382343862 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1311@172.16.200.35:5060>
Content-Length: 0

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

 
     -- Executing [1311@internal:1] e[1;36mDiale[0m("e[1;35mSIP/1310-0000006ae[0m", "e[1;35mSIP/1311,20e[0m") in new stack

    == Using SIP RTP CoS mark 5
 Audio is at 28704
 Adding codec 100004 (alaw) to SDP
 Adding codec 100008 (g729) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 Reliably Transmitting (no NAT) to 172.16.201.20:5060:
INVITE sip:1311@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK78d1b650
Max-Forwards: 70
From: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
To: <sip:1311@172.16.201.20;transport=udp>
Contact: <sip:1310@172.16.200.35:5060>
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 31 Dec 2014 01:27:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 617328596 617328596 IN IP4 172.16.200.35
s=Asterisk PBX 11.7.0
c=IN IP4 172.16.200.35
t=0 0
m=audio 28704 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

      -- Called SIP/1311
 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK78d1b650
From: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
To: <sip:1311@172.16.201.20;transport=udp>;tag=866759830
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Contact: 1311 <sip:1311@172.16.201.20;transport=udp>
Content-Length: 0

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

 --- (9 headers 0 lines) ---

 list_route: hop: <sip:1311@172.16.201.20;transport=udp>

     -- SIP/1311-0000006b is ringing
 
<--- Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKed92214e821c72335;received=172.16.201.20
From: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
To: 1311 <sip:1311@172.16.200.35:5060>;tag=as67f3b66b
Call-ID: 57e938dda8136fac
CSeq: 382343862 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1311@172.16.200.35:5060>
Content-Length: 0

<------------>
 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK78d1b650
From: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
To: <sip:1311@172.16.201.20;transport=udp>;tag=866759830
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Contact: 1311 <sip:1311@172.16.201.20;transport=udp>
Supported: timer, replaces
Content-Type: application/sdp
Content-Length: 295

v=0
o=MxSIP 0 0 IN IP4 172.16.201.20
s=SIP Call
c=IN IP4 172.16.201.20
t=0 0
m=audio 53460 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=yes
a=fmtp:101 0-15,144,149,159
a=ptime:20
a=sendrecv
a=silenceSupp:on - - - -
<------------->
 
 --- (11 headers 14 lines) ---
 Found RTP audio format 8
 Found RTP audio format 18
 Found RTP audio format 101
 Found audio description format PCMA for ID 8
 Found audio description format G729 for ID 18
 Found audio description format telephone-event for ID 101

 Capabilities: us - (alaw|g729), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

 Peer audio RTP is at port 172.16.201.20:53460
 
 list_route: hop: <sip:1311@172.16.201.20;transport=udp>
 set_destination: Parsing <sip:1311@172.16.201.20;transport=udp> for address/port to send to
 set_destination: set destination to 172.16.201.20:5060

 Transmitting (no NAT) to 172.16.201.20:5060:
ACK sip:1311@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK3591098e
Max-Forwards: 70
From: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
To: <sip:1311@172.16.201.20;transport=udp>;tag=866759830
Contact: <sip:1310@172.16.200.35:5060>
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0

---
 
     -- SIP/1311-0000006b answered SIP/1310-0000006a

 Audio is at 28086
 Adding codec 100004 (alaw) to SDP
 Adding codec 100008 (g729) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 
<--- Reliably Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKed92214e821c72335;received=172.16.201.20
From: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
To: 1311 <sip:1311@172.16.200.35:5060>;tag=as67f3b66b
Call-ID: 57e938dda8136fac
CSeq: 382343862 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1311@172.16.200.35:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 442804290 442804290 IN IP4 172.16.200.35
s=Asterisk PBX 11.7.0
c=IN IP4 172.16.200.35
t=0 0
m=audio 28086 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

     -- Remotely bridging SIP/1310-0000006a and SIP/1311-0000006b

 set_destination: Parsing <sip:1311@172.16.201.20;transport=udp> for address/port to send to
 set_destination: set destination to 172.16.201.20:5060

 Audio is at 28704
 Adding codec 100004 (alaw) to SDP
 Adding codec 100008 (g729) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 Reliably Transmitting (no NAT) to 172.16.201.20:5060:
INVITE sip:1311@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK362769e4
Max-Forwards: 70
From: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
To: <sip:1311@172.16.201.20;transport=udp>;tag=866759830
Contact: <sip:1310@172.16.200.35:5060>
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 617328596 617328597 IN IP4 172.16.201.20
s=Asterisk PBX 11.7.0
c=IN IP4 172.16.201.20
t=0 0
m=audio 53458 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
 
<--- SIP read from UDP:172.16.201.20:5060 --->
BYE sip:1310@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bK5facf6c3fbce3ff90
Max-Forwards: 70
From: <sip:1311@172.16.201.20;transport=udp>;tag=866759830
To: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 1757665239 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Supported: timer
User-Agent: Asterisk
Content-Length: 0

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

 --- (11 headers 0 lines) ---

 Sending to 172.16.201.20:5060 (no NAT)
 Scheduling destruction of SIP dialog '07edeefc3baf7f24540189d24011d460@172.16.200.35:5060' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.16.201.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bK5facf6c3fbce3ff90;received=172.16.201.20
From: <sip:1311@172.16.201.20;transport=udp>;tag=866759830
To: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 1757665239 BYE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

   == Spawn extension (internal, 1311, 1) exited non-zero on 'SIP/1310-0000006a'
 
 Scheduling destruction of SIP dialog '57e938dda8136fac' in 6400 ms (Method: INVITE)
 
 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK362769e4
From: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
To: <sip:1311@172.16.201.20;transport=udp>;tag=866759830
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Content-Length: 0

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

 --- (8 headers 0 lines) ---
 set_destination: Parsing <sip:1311@172.16.201.20;transport=udp> for address/port to send to
 set_destination: set destination to 172.16.201.20:5060

 Transmitting (no NAT) to 172.16.201.20:5060:
ACK sip:1311@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK362769e4
Max-Forwards: 70
From: <sip:1311@172.16.201.20;transport=udp>;tag=866759830
To: "1310" <sip:1310@172.16.200.35>;tag=as48609ca7
Contact: <sip:1310@172.16.200.35:5060>
Call-ID: 07edeefc3baf7f24540189d24011d460@172.16.200.35:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0

---
 
<--- SIP read from UDP:172.16.201.20:5060 --->
ACK sip:1311@172.16.200.35:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.201.20;branch=z9hG4bKf388472edf5d66e47
Max-Forwards: 70
From: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
To: 1311 <sip:1311@172.16.200.35:5060>;tag=as67f3b66b
Call-ID: 57e938dda8136fac
CSeq: 382343862 ACK
Authorization: Digest username="1310",realm="asterisk.rng.com",nonce="0d0916e3",uri="sip:1311@172.16.200.35:5060",response="f53a3f7c991170b607f702d6319bd9db",algorithm=MD5
User-Agent: Asterisk
Content-Length: 0

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

 --- (10 headers 0 lines) ---

 set_destination: Parsing <sip:1310@172.16.201.20;transport=udp> for address/port to send to
 set_destination: set destination to 172.16.201.20:5060

 Reliably Transmitting (no NAT) to 172.16.201.20:5060:
BYE sip:1310@172.16.201.20;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK5d3c2e80
Max-Forwards: 70
From: 1311 <sip:1311@172.16.200.35:5060>;tag=as67f3b66b
To: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
Call-ID: 57e938dda8136fac
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.7.0
Proxy-Authorization: Digest username="1310", realm="asterisk.rng.com", algorithm=MD5, uri="sip:172.16.200.35", nonce="0d0916e3", response="d609959e7c339de8dbc60fb7b36693d6"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

 Scheduling destruction of SIP dialog '57e938dda8136fac' in 6400 ms (Method: ACK)
 
<--- SIP read from UDP:172.16.201.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.200.35:5060;branch=z9hG4bK5d3c2e80
From: 1311 <sip:1311@172.16.200.35:5060>;tag=as67f3b66b
To: 1310 <sip:1310@172.16.200.35>;tag=09c63f525b
Call-ID: 57e938dda8136fac
CSeq: 102 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO
Content-Length: 0

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

 --- (8 headers 0 lines) ---

 SIP Response message for INCOMING dialog BYE arrived
 Really destroying SIP dialog '57e938dda8136fac' Method: ACK
 Really destroying SIP dialog '07edeefc3baf7f24540189d24011d460@172.16.200.35:5060' Method: BYE

 exit

Asterisk cleanly ending (0).
Executing last minute cleanups

The called device dropped the call immediately after it answered it. Whilst there are circumstances were a calling device can do that, there is not good reason for a called device to do that, as it can reject the INVITE, instead.

Maybe it discovered, too late, that it doesn’t support G.729, but it should not have accepted the offer of G.729 if that was the case.

(The call was actually up for a fraction of a second (although, as you screen-scraped, rather than used the log files, I’m relying on the fact that it collided with a re-invite to deduce the timing.)

The called device might have broken re-invite handing, so you could try disabling directmedia and sendrpid, but it was not behaving in a way consistent with the proper rejection of a re-invite.

Hi,

Thanks for your time and valuable reply but in the mean time we have resolved the issue.
we had some DQoS related configuration problem in our packet cable setup, in which the CMTS was not able to allocate bandwidth.