[asterisk-bugs] [JIRA] (ASTERISK-26637) chan_sip: Video TLS SRTP Broken

John Doe (JIRA) noreply at issues.asterisk.org
Mon Dec 5 12:28:14 CST 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26637?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234056#comment-234056 ] 

John Doe commented on ASTERISK-26637:
-------------------------------------

1) Disallow invites for both endpoints to force entire duration of call through asterisk pbx.

2) Enable TLS signalling over port 5061 with certificates from a trusted certificate authority.

2) Enable SRTP media encryption.

3) Initiate an audio call between two endpoints with video capability.

4) Shortly after call has been established, enabling video feed on each endpoint issues the error message mentioned above.

See detailed debug below:

cloud4*CLI> sip set debug peer 1001
SIP Debugging Enabled for IP: 192.168.74.70
[2016-12-02 22:49:56] NOTICE[3786]: chan_sip.c:15601 sip_reregister:    -- Re-registration for  XXXXXXXXXXX at callcentric.com
[2016-12-02 22:49:56] NOTICE[3786]: chan_sip.c:24394 handle_response_register: Outbound Registration: Expiry for callcentric.com is 60 sec (Scheduling reregistration in 45 s)

<--- SIP read from TLS:192.168.74.70:33059 --->
INVITE sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---f16597fb73c27ff9;rport
Max-Forwards: 70
Contact: <sip:1001 at 192.168.74.70:33059;transport=TLS>
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: Zoiper rv2.8.15
Allow-Events: presence, kpml, talk
Content-Length: 773

v=0
o=Zoiper 0 0 IN IP4 192.168.74.70
s=Zoiper
c=IN IP4 192.168.74.70
t=0 0
m=audio 58466 RTP/SAVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIflkHvR2mEJaoGyMF+EmlJe7A==
a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIflkHvR2mEJaoGyMF+EmlJe7A==
a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIflkHvR2mEJaoE=
a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIflkHvR2mEJaoE=
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIfl
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIfl
<------------->
--- (12 headers 16 lines) ---
Sending to 192.168.74.70:33059 (no NAT)
Sending to 192.168.74.70:33059 (no NAT)
Using INVITE request as basis request - -1YlDayNRP8XGrf4hfJ47w..
Found peer '1001' for '1001' from 192.168.74.70:33059

<--- Reliably Transmitting (NAT) to 192.168.74.70:33059 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---f16597fb73c27ff9;received=192.168.74.70;rport=33059
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as5d8df09b
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 1 INVITE
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="365457f2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '-1YlDayNRP8XGrf4hfJ47w..' in 6400 ms (Method: INVITE)

<--- SIP read from TLS:192.168.74.70:33059 --->
ACK sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---f16597fb73c27ff9;rport
Max-Forwards: 70
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as5d8df09b
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from TLS:192.168.74.70:33059 --->
INVITE sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---551b1274c9885cd9;rport
Max-Forwards: 70
Contact: <sip:1001 at 192.168.74.70:33059;transport=TLS>
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 2 INVITE
Content-Type: application/sdp
User-Agent: Zoiper rv2.8.15
Authorization: Digest username="1001",realm="asterisk",nonce="365457f2",uri="sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS",response="38c7e89ea0b1329f9839c131a5afcf20",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 773

v=0
o=Zoiper 0 0 IN IP4 192.168.74.70
s=Zoiper
c=IN IP4 192.168.74.70
t=0 0
m=audio 58466 RTP/SAVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIflkHvR2mEJaoGyMF+EmlJe7A==
a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIflkHvR2mEJaoGyMF+EmlJe7A==
a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIflkHvR2mEJaoE=
a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIflkHvR2mEJaoE=
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIfl
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIfl
<------------->
--- (13 headers 16 lines) ---
Sending to 192.168.74.70:33059 (NAT)
Using INVITE request as basis request - -1YlDayNRP8XGrf4hfJ47w..
Found peer '1001' for '1001' from 192.168.74.70:33059
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
[2016-12-02 22:50:07] WARNING[15710][C-00000011]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_256_CM_HMAC_SHA1_80
[2016-12-02 22:50:07] WARNING[15710][C-00000011]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_256_CM_HMAC_SHA1_32
[2016-12-02 22:50:07] WARNING[15710][C-00000011]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_192_CM_HMAC_SHA1_80
[2016-12-02 22:50:07] WARNING[15710][C-00000011]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_192_CM_HMAC_SHA1_32
Capabilities: us - (g722|ulaw|g729|h264), peer - audio=(g722)/video=(nothing)/text=(nothing), combined - (g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.74.70:58466
Peer doesn't provide video
Looking for 1013 in from-internal (domain xxxxxx.xxxxxxxxxxxxxx.com)
sip_route_dump: route/path hop: <sip:1001 at 192.168.74.70:33059;transport=TLS>

<--- Transmitting (NAT) to 192.168.74.70:33059 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---551b1274c9885cd9;received=192.168.74.70;rport=33059
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 2 INVITE
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1013 at XXX.XXX.XXX.XXX:5061;transport=TLS>
Content-Length: 0


<------------>
    -- Executing [1013 at from-internal:1] GotoIf("SIP/1001-00000020", "1?ext-local,1013,1:followme-check,1013,1") in new stack
    -- Goto (ext-local,1013,1)
    -- Executing [1013 at ext-local:1] Set("SIP/1001-00000020", "__RINGTIMER=15") in new stack
    -- Executing [1013 at ext-local:2] Macro("SIP/1001-00000020", "exten-vm,novm,1013,0,0,0") in new stack
    -- Executing [s at macro-exten-vm:1] Macro("SIP/1001-00000020", "user-callerid,") in new stack
    -- Executing [s at macro-user-callerid:1] Set("SIP/1001-00000020", "TOUCH_MONITOR=1480737007.35") in new stack
    -- Executing [s at macro-user-callerid:2] Set("SIP/1001-00000020", "AMPUSER=1001") in new stack
    -- Executing [s at macro-user-callerid:3] GotoIf("SIP/1001-00000020", "0?report") in new stack
    -- Executing [s at macro-user-callerid:4] ExecIf("SIP/1001-00000020", "1?Set(REALCALLERIDNUM=1001)") in new stack
    -- Executing [s at macro-user-callerid:5] Set("SIP/1001-00000020", "AMPUSER=1001") in new stack
    -- Executing [s at macro-user-callerid:6] GotoIf("SIP/1001-00000020", "0?limit") in new stack
    -- Executing [s at macro-user-callerid:7] Set("SIP/1001-00000020", "AMPUSERCIDNAME=JOHN DOE") in new stack
    -- Executing [s at macro-user-callerid:8] GotoIf("SIP/1001-00000020", "0?report") in new stack
    -- Executing [s at macro-user-callerid:9] Set("SIP/1001-00000020", "AMPUSERCID=1001") in new stack
    -- Executing [s at macro-user-callerid:10] Set("SIP/1001-00000020", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [s at macro-user-callerid:11] Set("SIP/1001-00000020", "CALLERID(all)="JOHN DOE" <1001>") in new stack
    -- Executing [s at macro-user-callerid:12] GotoIf("SIP/1001-00000020", "0?limit") in new stack
    -- Executing [s at macro-user-callerid:13] ExecIf("SIP/1001-00000020", "0?Set(GROUP(concurrency_limit)=1001)") in new stack
    -- Executing [s at macro-user-callerid:14] ExecIf("SIP/1001-00000020", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s at macro-user-callerid:15] GotoIf("SIP/1001-00000020", "0?continue") in new stack
    -- Executing [s at macro-user-callerid:16] ExecIf("SIP/1001-00000020", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s at macro-user-callerid:17] Set("SIP/1001-00000020", "__TTL=64") in new stack
    -- Executing [s at macro-user-callerid:18] GotoIf("SIP/1001-00000020", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [s at macro-user-callerid:29] Set("SIP/1001-00000020", "CALLERID(number)=1001") in new stack
    -- Executing [s at macro-user-callerid:30] Set("SIP/1001-00000020", "CALLERID(name)=JOHN DOE") in new stack
    -- Executing [s at macro-user-callerid:31] GotoIf("SIP/1001-00000020", "0?cnum") in new stack
    -- Executing [s at macro-user-callerid:32] Set("SIP/1001-00000020", "CDR(cnam)=JOHN DOE") in new stack
    -- Executing [s at macro-user-callerid:33] Set("SIP/1001-00000020", "CDR(cnum)=1001") in new stack
    -- Executing [s at macro-user-callerid:34] Set("SIP/1001-00000020", "CHANNEL(language)=en") in new stack
    -- Executing [s at macro-exten-vm:2] Set("SIP/1001-00000020", "RingGroupMethod=none") in new stack
    -- Executing [s at macro-exten-vm:3] Set("SIP/1001-00000020", "__EXTTOCALL=1013") in new stack
    -- Executing [s at macro-exten-vm:4] Set("SIP/1001-00000020", "__PICKUPMARK=1013") in new stack
    -- Executing [s at macro-exten-vm:5] Set("SIP/1001-00000020", "RT=") in new stack
    -- Executing [s at macro-exten-vm:6] ExecIf("SIP/1001-00000020", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
    -- Executing [s at macro-exten-vm:7] ExecIf("SIP/1001-00000020", "0?MacroExit()") in new stack
    -- Executing [s at macro-exten-vm:8] Gosub("SIP/1001-00000020", "sub-record-check,s,1(exten,1013,dontcare)") in new stack
    -- Executing [s at sub-record-check:1] GotoIf("SIP/1001-00000020", "0?initialized") in new stack
    -- Executing [s at sub-record-check:2] Set("SIP/1001-00000020", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s at sub-record-check:3] Set("SIP/1001-00000020", "NOW=1480737007") in new stack
    -- Executing [s at sub-record-check:4] Set("SIP/1001-00000020", "__DAY=02") in new stack
    -- Executing [s at sub-record-check:5] Set("SIP/1001-00000020", "__MONTH=12") in new stack
    -- Executing [s at sub-record-check:6] Set("SIP/1001-00000020", "__YEAR=2016") in new stack
    -- Executing [s at sub-record-check:7] Set("SIP/1001-00000020", "__TIMESTR=20161202-225007") in new stack
    -- Executing [s at sub-record-check:8] Set("SIP/1001-00000020", "__FROMEXTEN=1001") in new stack
    -- Executing [s at sub-record-check:9] Set("SIP/1001-00000020", "__MON_FMT=wav") in new stack
    -- Executing [s at sub-record-check:10] NoOp("SIP/1001-00000020", "Recordings initialized") in new stack
    -- Executing [s at sub-record-check:11] ExecIf("SIP/1001-00000020", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s at sub-record-check:12] Set("SIP/1001-00000020", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s at sub-record-check:13] ExecIf("SIP/1001-00000020", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s at sub-record-check:14] GotoIf("SIP/1001-00000020", "5?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s at sub-record-check:17] GotoIf("SIP/1001-00000020", "1?sub-record-check,exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten at sub-record-check:1] NoOp("SIP/1001-00000020", "Exten Recording Check between 1001 and 1013") in new stack
    -- Executing [exten at sub-record-check:2] Set("SIP/1001-00000020", "CALLTYPE=internal") in new stack
    -- Executing [exten at sub-record-check:3] ExecIf("SIP/1001-00000020", "0?Set(CALLTYPE=)") in new stack
    -- Executing [exten at sub-record-check:4] Set("SIP/1001-00000020", "CALLEE=dontcare") in new stack
    -- Executing [exten at sub-record-check:5] ExecIf("SIP/1001-00000020", "0?Set(CALLEE=dontcare)") in new stack
    -- Executing [exten at sub-record-check:6] GotoIf("SIP/1001-00000020", "0?callee") in new stack
    -- Executing [exten at sub-record-check:7] GotoIf("SIP/1001-00000020", "1?caller") in new stack
    -- Goto (sub-record-check,exten,13)
    -- Executing [exten at sub-record-check:13] Set("SIP/1001-00000020", "RECMODE=force") in new stack
    -- Executing [exten at sub-record-check:14] ExecIf("SIP/1001-00000020", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten at sub-record-check:15] ExecIf("SIP/1001-00000020", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten at sub-record-check:16] Gosub("SIP/1001-00000020", "recordcheck,1(force,internal,1013)") in new stack
    -- Executing [recordcheck at sub-record-check:1] NoOp("SIP/1001-00000020", "Starting recording check against force") in new stack
    -- Executing [recordcheck at sub-record-check:2] Goto("SIP/1001-00000020", "force") in new stack
    -- Goto (sub-record-check,recordcheck,5)
    -- Executing [recordcheck at sub-record-check:5] Set("SIP/1001-00000020", "__REC_POLICY_MODE=FORCE") in new stack
    -- Executing [recordcheck at sub-record-check:6] GotoIf("SIP/1001-00000020", "1?startrec") in new stack
    -- Goto (sub-record-check,recordcheck,16)
    -- Executing [recordcheck at sub-record-check:16] NoOp("SIP/1001-00000020", "Starting recording: internal, 1013") in new stack
    -- Executing [recordcheck at sub-record-check:17] Set("SIP/1001-00000020", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
    -- Executing [recordcheck at sub-record-check:18] Set("SIP/1001-00000020", "__CALLFILENAME=internal-1013-1001-20161202-225007-1480737007.35") in new stack
    -- Executing [recordcheck at sub-record-check:19] MixMonitor("SIP/1001-00000020", "2016/12/02/internal-1013-1001-20161202-225007-1480737007.35.wav,ai(LOCAL_MIXMON_ID),") in new stack
    -- Executing [recordcheck at sub-record-check:20] Set("SIP/1001-00000020", "__MIXMON_ID=0x7ff6f0071410") in new stack
    -- Executing [recordcheck at sub-record-check:21] Set("SIP/1001-00000020", "__RECORD_ID=SIP/1001-00000020") in new stack
    -- Executing [recordcheck at sub-record-check:22] Set("SIP/1001-00000020", "__REC_STATUS=RECORDING") in new stack
    -- Executing [recordcheck at sub-record-check:23] Set("SIP/1001-00000020", "CDR(recordingfile)=internal-1013-1001-20161202-225007-1480737007.35.wav") in new stack
    -- Executing [recordcheck at sub-record-check:24] Return("SIP/1001-00000020", "") in new stack
    -- Executing [exten at sub-record-check:17] Return("SIP/1001-00000020", "") in new stack
    -- Executing [s at macro-exten-vm:9] GotoIf("SIP/1001-00000020", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,15)
    -- Executing [s at macro-exten-vm:15] GosubIf("SIP/1001-00000020", "0?clrheader,1()") in new stack
    -- Executing [s at macro-exten-vm:16] Macro("SIP/1001-00000020", "dial-one,,Ttr,1013") in new stack
    -- Executing [s at macro-dial-one:1] Set("SIP/1001-00000020", "DEXTEN=1013") in new stack
    -- Executing [s at macro-dial-one:2] ExecIf("SIP/1001-00000020", "0?Set(EXTTOCALL=1013)") in new stack
    -- Executing [s at macro-dial-one:3] Set("SIP/1001-00000020", "DIALSTATUS_CW=") in new stack
    -- Executing [s at macro-dial-one:4] GosubIf("SIP/1001-00000020", "0?screen,1()") in new stack
    -- Executing [s at macro-dial-one:5] GosubIf("SIP/1001-00000020", "0?cf,1()") in new stack
    -- Executing [s at macro-dial-one:6] GotoIf("SIP/1001-00000020", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,9)
    -- Executing [s at macro-dial-one:9] GotoIf("SIP/1001-00000020", "0?nodial") in new stack
    -- Executing [s at macro-dial-one:10] GotoIf("SIP/1001-00000020", "0?continue") in new stack
    -- Executing [s at macro-dial-one:11] Set("SIP/1001-00000020", "EXTHASCW=ENABLED") in new stack
    -- Executing [s at macro-dial-one:12] GotoIf("SIP/1001-00000020", "0?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,24)
    -- Executing [s at macro-dial-one:24] GotoIf("SIP/1001-00000020", "0?next3:continue") in new stack
    -- Goto (macro-dial-one,s,26)
    -- Executing [s at macro-dial-one:26] GotoIf("SIP/1001-00000020", "0?nodial") in new stack
    -- Executing [s at macro-dial-one:27] GosubIf("SIP/1001-00000020", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [dstring at macro-dial-one:1] Set("SIP/1001-00000020", "DSTRING=") in new stack
    -- Executing [dstring at macro-dial-one:2] Set("SIP/1001-00000020", "DEVICES=1013") in new stack
    -- Executing [dstring at macro-dial-one:3] ExecIf("SIP/1001-00000020", "0?Return()") in new stack
    -- Executing [dstring at macro-dial-one:4] ExecIf("SIP/1001-00000020", "0?Set(DEVICES=013)") in new stack
    -- Executing [dstring at macro-dial-one:5] Set("SIP/1001-00000020", "LOOPCNT=1") in new stack
    -- Executing [dstring at macro-dial-one:6] Set("SIP/1001-00000020", "ITER=1") in new stack
    -- Executing [dstring at macro-dial-one:7] Set("SIP/1001-00000020", "THISDIAL=SIP/1013") in new stack
    -- Executing [dstring at macro-dial-one:8] GosubIf("SIP/1001-00000020", "1?zap2dahdi,1()") in new stack
  == Begin MixMonitor Recording SIP/1001-00000020
    -- Executing [zap2dahdi at macro-dial-one:1] ExecIf("SIP/1001-00000020", "0?Return()") in new stack
    -- Executing [zap2dahdi at macro-dial-one:2] Set("SIP/1001-00000020", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi at macro-dial-one:3] Set("SIP/1001-00000020", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi at macro-dial-one:4] Set("SIP/1001-00000020", "ITER2=1") in new stack
    -- Executing [zap2dahdi at macro-dial-one:5] Set("SIP/1001-00000020", "THISPART2=SIP/1013") in new stack
    -- Executing [zap2dahdi at macro-dial-one:6] ExecIf("SIP/1001-00000020", "0?Set(THISPART2=DAHDI/1013)") in new stack
    -- Executing [zap2dahdi at macro-dial-one:7] Set("SIP/1001-00000020", "NEWDIAL=SIP/1013&") in new stack
    -- Executing [zap2dahdi at macro-dial-one:8] Set("SIP/1001-00000020", "ITER2=2") in new stack
    -- Executing [zap2dahdi at macro-dial-one:9] GotoIf("SIP/1001-00000020", "0?begin2") in new stack
    -- Executing [zap2dahdi at macro-dial-one:10] Set("SIP/1001-00000020", "THISDIAL=SIP/1013") in new stack
    -- Executing [zap2dahdi at macro-dial-one:11] Return("SIP/1001-00000020", "") in new stack
    -- Executing [dstring at macro-dial-one:9] GotoIf("SIP/1001-00000020", "1?docheck") in new stack
    -- Goto (macro-dial-one,dstring,14)
    -- Executing [dstring at macro-dial-one:14] GotoIf("SIP/1001-00000020", "0?skipset") in new stack
    -- Executing [dstring at macro-dial-one:15] Set("SIP/1001-00000020", "DSTRING=SIP/1013&") in new stack
    -- Executing [dstring at macro-dial-one:16] Set("SIP/1001-00000020", "ITER=2") in new stack
    -- Executing [dstring at macro-dial-one:17] GotoIf("SIP/1001-00000020", "0?begin") in new stack
    -- Executing [dstring at macro-dial-one:18] ExecIf("SIP/1001-00000020", "0?Return()") in new stack
    -- Executing [dstring at macro-dial-one:19] Set("SIP/1001-00000020", "DSTRING=SIP/1013") in new stack
    -- Executing [dstring at macro-dial-one:20] Return("SIP/1001-00000020", "") in new stack
    -- Executing [s at macro-dial-one:28] GotoIf("SIP/1001-00000020", "0?nodial") in new stack
    -- Executing [s at macro-dial-one:29] GotoIf("SIP/1001-00000020", "0?skiptrace") in new stack
    -- Executing [s at macro-dial-one:30] GosubIf("SIP/1001-00000020", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset at macro-dial-one:1] Set("SIP/1001-00000020", "DB(CALLTRACE/1013)=1001") in new stack
    -- Executing [ctset at macro-dial-one:2] Return("SIP/1001-00000020", "") in new stack
    -- Executing [s at macro-dial-one:31] Set("SIP/1001-00000020", "D_OPTIONS=Ttr") in new stack
    -- Executing [s at macro-dial-one:32] NoOp("SIP/1001-00000020", "Blind Transfer: , Attended Transfer: , User: 1001, Alert Info: ") in new stack
    -- Executing [s at macro-dial-one:33] ExecIf("SIP/1001-00000020", "1?Set(ALERT_INFO=)") in new stack
    -- Executing [s at macro-dial-one:34] ExecIf("SIP/1001-00000020", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s at macro-dial-one:35] ExecIf("SIP/1001-00000020", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s at macro-dial-one:36] ExecIf("SIP/1001-00000020", "0?Set(ALERT_INFO=;volume=)") in new stack
    -- Executing [s at macro-dial-one:37] ExecIf("SIP/1001-00000020", "0?Set(ALERT_INFO=;volume=)") in new stack
    -- Executing [s at macro-dial-one:38] GosubIf("SIP/1001-00000020", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
    -- Executing [s at macro-dial-one:39] ExecIf("SIP/1001-00000020", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s at macro-dial-one:40] GosubIf("SIP/1001-00000020", "0?qwait,1()") in new stack
    -- Executing [s at macro-dial-one:41] Set("SIP/1001-00000020", "__CWIGNORE=") in new stack
    -- Executing [s at macro-dial-one:42] Set("SIP/1001-00000020", "__KEEPCID=TRUE") in new stack
    -- Executing [s at macro-dial-one:43] GotoIf("SIP/1001-00000020", "0?usegoto,1") in new stack
    -- Executing [s at macro-dial-one:44] GotoIf("SIP/1001-00000020", "0?godial") in new stack
    -- Executing [s at macro-dial-one:45] Gosub("SIP/1001-00000020", "sub-presencestate-display,s,1(1013)") in new stack
    -- Executing [s at sub-presencestate-display:1] Goto("SIP/1001-00000020", "state-not_set,1") in new stack
    -- Goto (sub-presencestate-display,state-not_set,1)
    -- Executing [state-not_set at sub-presencestate-display:1] Set("SIP/1001-00000020", "PRESENCESTATE_DISPLAY=") in new stack
    -- Executing [state-not_set at sub-presencestate-display:2] Return("SIP/1001-00000020", "") in new stack
    -- Executing [s at macro-dial-one:46] Set("SIP/1001-00000020", "CONNECTEDLINE(name,i)=JANE DOE") in new stack
    -- Executing [s at macro-dial-one:47] Set("SIP/1001-00000020", "CONNECTEDLINE(num)=1013") in new stack
    -- Executing [s at macro-dial-one:48] Set("SIP/1001-00000020", "D_OPTIONS=TtrI") in new stack
    -- Executing [s at macro-dial-one:49] Macro("SIP/1001-00000020", "dialout-one-predial-hook,") in new stack
    -- Executing [s at macro-dialout-one-predial-hook:1] MacroExit("SIP/1001-00000020", "") in new stack
    -- Executing [s at macro-dial-one:50] ExecIf("SIP/1001-00000020", "0?Set(D_OPTIONS=trII)") in new stack
    -- Executing [s at macro-dial-one:51] Dial("SIP/1001-00000020", "SIP/1013,,TtrIb(func-apply-sipheaders^s^1)") in new stack
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- SIP/1013-00000021 Internal Gosub(func-apply-sipheaders,s,1) start
    -- Executing [s at func-apply-sipheaders:1] NoOp("SIP/1013-00000021", "Applying SIP Headers to channel") in new stack
    -- Executing [s at func-apply-sipheaders:2] Set("SIP/1013-00000021", "SIPHEADERKEYS=") in new stack
    -- Executing [s at func-apply-sipheaders:3] While("SIP/1013-00000021", "0") in new stack
    -- Jumping to priority 7
    -- Executing [s at func-apply-sipheaders:8] Return("SIP/1013-00000021", "") in new stack
  == Spawn extension (from-internal, 1013, 1) exited non-zero on 'SIP/1013-00000021'
    -- SIP/1013-00000021 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
    -- Called SIP/1013

<--- Transmitting (NAT) to 192.168.74.70:33059 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---551b1274c9885cd9;received=192.168.74.70;rport=33059
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 2 INVITE
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1013 at XXX.XXX.XXX.XXX:5061;transport=TLS>
P-Asserted-Identity: "JANE DOE" <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com>
Content-Length: 0


<------------>
    -- Connected line update to SIP/1001-00000020 prevented.
    -- SIP/1013-00000021 is ringing

<--- Transmitting (NAT) to 192.168.74.70:33059 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---551b1274c9885cd9;received=192.168.74.70;rport=33059
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 2 INVITE
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1013 at XXX.XXX.XXX.XXX:5061;transport=TLS>
Content-Length: 0


<------------>
[2016-12-02 22:50:12] NOTICE[3786]: chan_sip.c:15601 sip_reregister:    -- Re-registration for  193909 at toronto4.voip.ms
[2016-12-02 22:50:12] NOTICE[3786]: chan_sip.c:24394 handle_response_register: Outbound Registration: Expiry for toronto4.voip.ms is 120 sec (Scheduling reregistration in 105 s)
       > 0x7ff6f00a97b0 -- Probation passed - setting RTP source address to 192.168.74.20:51378
    -- Connected line update to SIP/1001-00000020 prevented.
    -- SIP/1013-00000021 answered SIP/1001-00000020
Audio is at 14628
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.74.70:33059 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---551b1274c9885cd9;received=192.168.74.70;rport=33059
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 2 INVITE
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1013 at XXX.XXX.XXX.XXX:5061;transport=TLS>
P-Asserted-Identity: "JANE DOE" <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com>
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 255018580 255018580 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk PBX 13.12.2
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 14628 RTP/SAVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7UlPjruGDmqO6n4y9fp/SBKHGg05Xj1Yo48rvf16

<------------>
    -- Channel SIP/1013-00000021 joined 'simple_bridge' basic-bridge <95c37ce6-f835-4d88-9bf2-3707c009fe7d>
    -- Channel SIP/1001-00000020 joined 'simple_bridge' basic-bridge <95c37ce6-f835-4d88-9bf2-3707c009fe7d>
       > 0x7ff6f0093eb0 -- Probation passed - setting RTP source address to 192.168.74.70:58466

<--- SIP read from TLS:192.168.74.70:33059 --->
ACK sip:1013 at XXX.XXX.XXX.XXX:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---373e369e9554c16f;rport
Max-Forwards: 70
Contact: <sip:1001 at 192.168.74.70:33059;transport=TLS>
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 2 ACK
User-Agent: Zoiper rv2.8.15
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
       > 0x7ff6f00a97b0 -- Probation passed - setting RTP source address to 192.168.74.20:51378

<--- SIP read from TLS:192.168.74.70:33059 --->


<------------->
Really destroying SIP dialog 'DlH2Df_0Vvv79JY-A1Xxjw..' Method: REGISTER
Really destroying SIP dialog 'zwSDNpoCSsJIaWdWm5G2nw..' Method: REGISTER

<--- SIP read from TLS:192.168.74.70:33059 --->
INVITE sip:1013 at XXX.XXX.XXX.XXX:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---ecae9b3c03389e7f;rport
Max-Forwards: 70
Contact: <sip:1001 at 192.168.74.70:33059;transport=TLS>
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 3 INVITE
Content-Type: application/sdp
User-Agent: Zoiper rv2.8.15
Authorization: Digest username="1001",realm="asterisk",nonce="365457f2",uri="sip:1013 at XXX.XXX.XXX.XXX:5061;transport=TLS",response="1647e6f8f55b58dacf5d934898e8487a",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 922

v=0
o=Zoiper 0 1 IN IP4 192.168.74.70
s=Zoiper
c=IN IP4 192.168.74.70
t=0 0
m=audio 58466 RTP/SAVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RmhGJpDw6M0wrQoQebA4tMY61maZ0Lxry0sdBIfl
m=video 58468 RTP/SAVP 118
a=rtpmap:118 H264/90000
a=sendrecv
a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:ewzbIFxe1jtlTIdZtWRt16Dj8tz6LuFgQN2TLkAovLcp+BvocsNhv8QrP9hq/g==
a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:ewzbIFxe1jtlTIdZtWRt16Dj8tz6LuFgQN2TLkAovLcp+BvocsNhv8QrP9hq/g==
a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:ewzbIFxe1jtlTIdZtWRt16Dj8tz6LuFgQN2TLkAovLcp+BvocsM=
a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:ewzbIFxe1jtlTIdZtWRt16Dj8tz6LuFgQN2TLkAovLcp+BvocsM=
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ewzbIFxe1jtlTIdZtWRt16Dj8tz6LuFgQN2TLkAo
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:ewzbIFxe1jtlTIdZtWRt16Dj8tz6LuFgQN2TLkAo
<------------->
--- (13 headers 20 lines) ---
Sending to 192.168.74.70:33059 (NAT)
Found RTP audio format 9
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Found RTP video format 118
Found video description format H264 for ID 118
[2016-12-02 22:50:26] WARNING[15710][C-00000011]: chan_sip.c:10712 process_sdp: Rejecting secure video stream without encryption details: video 58468 RTP/SAVP 118

<--- Reliably Transmitting (NAT) to 192.168.74.70:33059 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---ecae9b3c03389e7f;received=192.168.74.70;rport=33059
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 3 INVITE
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


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

<--- SIP read from TLS:192.168.74.70:33059 --->
ACK sip:1013 at XXX.XXX.XXX.XXX:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---ecae9b3c03389e7f;rport
Max-Forwards: 70
To: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 3 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[2016-12-02 22:50:34] WARNING[15668][C-00000011]: chan_sip.c:10712 process_sdp: Rejecting secure video stream without encryption details: video 51380 RTP/SAVP 118
       > 0x7ff6f00a97b0 -- Probation passed - setting RTP source address to 192.168.74.20:51378
    -- Channel SIP/1013-00000021 left 'simple_bridge' basic-bridge <95c37ce6-f835-4d88-9bf2-3707c009fe7d>
    -- Channel SIP/1001-00000020 left 'simple_bridge' basic-bridge <95c37ce6-f835-4d88-9bf2-3707c009fe7d>
  == Spawn extension (macro-dial-one, s, 51) exited non-zero on 'SIP/1001-00000020' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 16) exited non-zero on 'SIP/1001-00000020' in macro 'exten-vm'
  == Spawn extension (ext-local, 1013, 2) exited non-zero on 'SIP/1001-00000020'
    -- Executing [h at ext-local:1] Macro("SIP/1001-00000020", "hangupcall,") in new stack
    -- Executing [s at macro-hangupcall:1] GotoIf("SIP/1001-00000020", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s at macro-hangupcall:3] ExecIf("SIP/1001-00000020", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s at macro-hangupcall:4] Hangup("SIP/1001-00000020", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/1001-00000020' in macro 'hangupcall'
  == Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/1001-00000020'
Scheduling destruction of SIP dialog '-1YlDayNRP8XGrf4hfJ47w..' in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 192.168.74.70:33059:
BYE sip:1001 at 192.168.74.70:33059;transport=TLS SIP/2.0
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK0e48adc4;rport
Max-Forwards: 70
From: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
To: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 102 BYE
User-Agent: FPBX-13.0.190.7(13.12.1)
Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sips:xxxxxx.xxxxxxxxxxxxxx.com", nonce="365457f2", response="1fa1e19e9f438440011420796c3e23e7"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/1001-00000020

<--- SIP read from TLS:192.168.74.70:33059 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK0e48adc4;rport=5061
Contact: <sip:1001 at 192.168.74.70:33059;transport=TLS>
To: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=037ff50e
From: <sip:1013 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as7be5cbbe
Call-ID: -1YlDayNRP8XGrf4hfJ47w..
CSeq: 102 BYE
User-Agent: Zoiper rv2.8.15
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '-1YlDayNRP8XGrf4hfJ47w..' Method: ACK
[2016-12-02 22:50:41] NOTICE[3786]: chan_sip.c:15601 sip_reregister:    -- Re-registration for  XXXXXXXXXXX at callcentric.com
[2016-12-02 22:50:41] NOTICE[3786]: chan_sip.c:24394 handle_response_register: Outbound Registration: Expiry for callcentric.com is 61 sec (Scheduling reregistration in 46 s)
Reliably Transmitting (NAT) to 192.168.74.70:33059:
OPTIONS sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1 SIP/2.0
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK43b29e68;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as68d97055
To: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>
Contact: <sip:Unknown at XXX.XXX.XXX.XXX:5061;transport=TLS>
Call-ID: 7cc39c225c867ce87f59e0e837192d3a at XXX.XXX.XXX.XXX:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Sat, 03 Dec 2016 03:50:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from TLS:192.168.74.70:33059 --->


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

<--- SIP read from TLS:192.168.74.70:33059 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK43b29e68;rport=5061
Contact: <sip:192.168.74.70:50207;transport=tls>
To: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>;tag=9789696e
From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as68d97055
Call-ID: 7cc39c225c867ce87f59e0e837192d3a at XXX.XXX.XXX.XXX:5061
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper rv2.8.15
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '7cc39c225c867ce87f59e0e837192d3a at XXX.XXX.XXX.XXX:5061' Method: OPTIONS

<--- SIP read from TLS:192.168.74.70:33059 --->


<------------->
[2016-12-02 22:51:27] NOTICE[3786]: chan_sip.c:15601 sip_reregister:    -- Re-registration for  XXXXXXXXXXX at callcentric.com
[2016-12-02 22:51:27] NOTICE[3786]: chan_sip.c:24394 handle_response_register: Outbound Registration: Expiry for callcentric.com is 62 sec (Scheduling reregistration in 47 s)

<--- SIP read from TLS:192.168.74.70:33059 --->
REGISTER sip:xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---1d8c4174d398121c;rport
Max-Forwards: 70
Contact: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>
To: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=35e23302
Call-ID: zwSDNpoCSsJIaWdWm5G2nw..
CSeq: 3 REGISTER
Expires: 120
User-Agent: Zoiper rv2.8.15
Authorization: Digest username="1001",realm="asterisk",nonce="0967e7da",uri="sip:xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS",response="c6ff5de5f41d3664989e15dcb11ea75c",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.74.70:33059 (no NAT)
Sending to 192.168.74.70:33059 (no NAT)

<--- Transmitting (NAT) to 192.168.74.70:33059 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---1d8c4174d398121c;received=192.168.74.70;rport=33059
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=35e23302
To: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as138d8377
Call-ID: zwSDNpoCSsJIaWdWm5G2nw..
CSeq: 3 REGISTER
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ad16cf7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'zwSDNpoCSsJIaWdWm5G2nw..' in 32000 ms (Method: REGISTER)

<--- SIP read from TLS:192.168.74.70:33059 --->
REGISTER sip:xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---da70d72e76ff5664;rport
Max-Forwards: 70
Contact: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>
To: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=35e23302
Call-ID: zwSDNpoCSsJIaWdWm5G2nw..
CSeq: 4 REGISTER
Expires: 120
User-Agent: Zoiper rv2.8.15
Authorization: Digest username="1001",realm="asterisk",nonce="5ad16cf7",uri="sip:xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS",response="68971aeed0081000c50af32727a5707e",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.74.70:33059 (NAT)
Reliably Transmitting (NAT) to 192.168.74.70:33059:
OPTIONS sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1 SIP/2.0
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK26caeef9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as6835358a
To: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>
Contact: <sip:Unknown at XXX.XXX.XXX.XXX:5061;transport=TLS>
Call-ID: 376354487ace4ace165280b2719ba77e at XXX.XXX.XXX.XXX:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Sat, 03 Dec 2016 03:51:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 192.168.74.70:33059 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.74.70:33059;branch=z9hG4bK-524287-1---da70d72e76ff5664;received=192.168.74.70;rport=33059
From: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=35e23302
To: <sip:1001 at xxxxxx.xxxxxxxxxxxxxx.com:5061;transport=TLS>;tag=as138d8377
Call-ID: zwSDNpoCSsJIaWdWm5G2nw..
CSeq: 4 REGISTER
Server: FPBX-13.0.190.7(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>;expires=120
Date: Sat, 03 Dec 2016 03:51:35 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2560207b55186624291de3713c8e4c76 at XXX.XXX.XXX.XXX:5061' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.74.70:33059:
NOTIFY sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1 SIP/2.0
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK41c4db96;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as5ea47839
To: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>
Contact: <sip:Unknown at XXX.XXX.XXX.XXX:5061;transport=TLS>
Call-ID: 2560207b55186624291de3713c8e4c76 at XXX.XXX.XXX.XXX:5061
CSeq: 102 NOTIFY
User-Agent: FPBX-13.0.190.7(13.12.1)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 104

Messages-Waiting: yes
Message-Account: sip:*97 at XXX.XXX.XXX.XXX;transport=TLS
Voice-Message: 4/0 (0/0)

---
Scheduling destruction of SIP dialog 'zwSDNpoCSsJIaWdWm5G2nw..' in 32000 ms (Method: REGISTER)

<--- SIP read from TLS:192.168.74.70:33059 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK26caeef9;rport=5061
Contact: <sip:192.168.74.70:50207;transport=tls>
To: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>;tag=bdf7276a
From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as6835358a
Call-ID: 376354487ace4ace165280b2719ba77e at XXX.XXX.XXX.XXX:5061
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper rv2.8.15
Allow-Events: presence, kpml, talk
Content-Length: 0

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

<--- SIP read from TLS:192.168.74.70:33059 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK41c4db96;rport=5061
Contact: <sip:192.168.74.70:50207;transport=tls>
To: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>;tag=a93b4676
From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as5ea47839
Call-ID: 2560207b55186624291de3713c8e4c76 at XXX.XXX.XXX.XXX:5061
CSeq: 102 NOTIFY
User-Agent: Zoiper rv2.8.15
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '376354487ace4ace165280b2719ba77e at XXX.XXX.XXX.XXX:5061' Method: OPTIONS
Really destroying SIP dialog '2560207b55186624291de3713c8e4c76 at XXX.XXX.XXX.XXX:5061' Method: NOTIFY
[2016-12-02 22:51:57] NOTICE[3786]: chan_sip.c:15601 sip_reregister:    -- Re-registration for  193909 at toronto4.voip.ms
[2016-12-02 22:51:57] NOTICE[3786]: chan_sip.c:24394 handle_response_register: Outbound Registration: Expiry for toronto4.voip.ms is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog 'zwSDNpoCSsJIaWdWm5G2nw..' Method: REGISTER

<--- SIP read from TLS:192.168.74.70:33059 --->


<------------->
[2016-12-02 22:52:14] NOTICE[3786]: chan_sip.c:15601 sip_reregister:    -- Re-registration for  XXXXXXXXXXX at callcentric.com
[2016-12-02 22:52:14] NOTICE[3786]: chan_sip.c:24394 handle_response_register: Outbound Registration: Expiry for callcentric.com is 63 sec (Scheduling reregistration in 48 s)
Reliably Transmitting (NAT) to 192.168.74.70:33059:
OPTIONS sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1 SIP/2.0
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK41929880;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as525ea883
To: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>
Contact: <sip:Unknown at XXX.XXX.XXX.XXX:5061;transport=TLS>
Call-ID: 5c72de9442dcea2229d741c222514143 at XXX.XXX.XXX.XXX:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(13.12.1)
Date: Sat, 03 Dec 2016 03:52:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from TLS:192.168.74.70:33059 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK41929880;rport=5061
Contact: <sip:192.168.74.70:50207;transport=tls>
To: <sip:1001 at 192.168.74.70:33059;transport=TLS;rinstance=631fc073f821dcc1>;tag=dee5133b
From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as525ea883
Call-ID: 5c72de9442dcea2229d741c222514143 at XXX.XXX.XXX.XXX:5061
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper rv2.8.15
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '5c72de9442dcea2229d741c222514143 at XXX.XXX.XXX.XXX:5061' Method: OPTIONS

<--- SIP read from TLS:192.168.74.70:33059 --->


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

> chan_sip: Video TLS SRTP Broken
> -------------------------------
>
>                 Key: ASTERISK-26637
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26637
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: 13.12.1, 13.12.2
>         Environment: Centos 6.6
>            Reporter: John Doe
>            Assignee: John Doe
>
> Seeing the following in my logs when attempting to initiate video during a call.
> [2016-11-12 20:38:35] VERBOSE[19511][C-00000019] bridge_channel.c: Channel SIP/1001-0000002c joined 'simple_bridge' basic-bridge <93ab1bd6-e342-4c08-a9f5-7074240d318e>
> [2016-11-12 20:38:38] WARNING[19472][C-00000019] chan_sip.c: Rejecting secure video stream without encryption details: video 35906 RTP/SAVP 118
> [2016-11-12 20:38:44] ERROR[18924] tcptls.c: SSL_shutdown() failed: 5
> [2016-11-12 20:38:53] VERBOSE[19511][C-00000019] bridge_channel.c: Channel SIP/1001-0000002c left 'simple_bridge' basic-bridge <93ab1bd6-e342-4c08-a9f5-7074240d318e>



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list