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