SIP Video Broken - TLS SRTP

I don’t use video too often but I just recently noticed that it has stopped working. Using FreePBX 13 Asterisk 13.11.2 latest.

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>

Any thoughts?

It looks like the phone is enforcing media encryption for the video call but Asterisk is not set up for it. Check the phone and turn off SRTP.

Asterisk is set up for SRTP. The audio portion of the call works fine. When either endpoint then enables a video stream I get that error message. The call signaling and audio continues without issue.

I am still on the same version of my endpoint software that I was on a few months ago where it was working ok.

Could you post the full SDP?

You mean from the Asterisk CLI? Or do I have to do a packet capture with Wireshark?

SIP debug at the cli will be good.

I accessed the asterisk CLI via “asterisk -rvvvvv”.

I set up the audio calls and then monitored when I pressed the video button on both endpoints. Only the following two lines appear:

Pressing video button on endpoint 1:

[2016-12-02 22:30:44] WARNING[13930][C-00000010]: chan_sip.c:10712 process_sdp: Rejecting secure video stream without encryption details: video 51380 RTP/SAVP 118

Pressing video button on endpoint 2:

[2016-12-02 22:30:49] WARNING[13928][C-00000010]: chan_sip.c:10712 process_sdp: Rejecting secure video stream without encryption details: video 53866 RTP/SAVP 118

Nothing else is printed.

sip set debug peer 1001 in order to get the sip invite that will have the full SDP

cloud4*CLI> sip set debug peer 1001                                                                                                                                          Unable to get IP address of peer '1001'
    -- Registered SIP '1001' at 192.168.74.70:33059
[2016-12-02 22:49:47] NOTICE[15710]: chan_sip.c:17157 check_auth: Correct auth, but based on stale nonce received from '<sip:[email protected]:5061;transport=TLS>;tag=7ac16948'
    -- Unregistered SIP '1001'
    -- Registered SIP '1001' at 192.168.74.70:33059
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  [email protected]
[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:[email protected]: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:[email protected]:33059;transport=TLS>
To: <sip:[email protected]:5061;transport=TLS>
From: <sip:[email protected]: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:[email protected]:5061;transport=TLS>;tag=037ff50e
To: <sip:[email protected]: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:[email protected]: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:[email protected]:5061;transport=TLS>;tag=as5d8df09b
From: <sip:[email protected]: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:[email protected]: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:[email protected]:33059;transport=TLS>
To: <sip:[email protected]:5061;transport=TLS>
From: <sip:[email protected]: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:[email protected]: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:[email protected]: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:[email protected]:5061;transport=TLS>;tag=037ff50e
To: <sip:[email protected]: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:[email protected]:5061;transport=TLS>
Content-Length: 0


<------------>
    -- Executing [[email protected]:1] GotoIf("SIP/1001-00000020", "1?ext-local,1013,1:followme-check,1013,1") in new stack
    -- Goto (ext-local,1013,1)
    -- Executing [[email protected]:1] Set("SIP/1001-00000020", "__RINGTIMER=15") in new stack
    -- Executing [[email protected]:2] Macro("SIP/1001-00000020", "exten-vm,novm,1013,0,0,0") in new stack
    -- Executing [[email protected]:1] Macro("SIP/1001-00000020", "user-callerid,") in new stack
    -- Executing [[email protected]:1] Set("SIP/1001-00000020", "TOUCH_MONITOR=1480737007.35") in new stack
    -- Executing [[email protected]:2] Set("SIP/1001-00000020", "AMPUSER=1001") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/1001-00000020", "0?report") in new stack
    -- Executing [[email protected]:4] ExecIf("SIP/1001-00000020", "1?Set(REALCALLERIDNUM=1001)") in new stack
    -- Executing [[email protected]:5] Set("SIP/1001-00000020", "AMPUSER=1001") in new stack
    -- Executing [[email protected]:6] GotoIf("SIP/1001-00000020", "0?limit") in new stack
    -- Executing [[email protected]:7] Set("SIP/1001-00000020", "AMPUSERCIDNAME=JOHN DOE") in new stack
    -- Executing [[email protected]:8] GotoIf("SIP/1001-00000020", "0?report") in new stack
    -- Executing [[email protected]:9] Set("SIP/1001-00000020", "AMPUSERCID=1001") in new stack
    -- Executing [[email protected]:10] Set("SIP/1001-00000020", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [[email protected]:11] Set("SIP/1001-00000020", "CALLERID(all)="JOHN DOE" <1001>") in new stack
    -- Executing [[email protected]:12] GotoIf("SIP/1001-00000020", "0?limit") in new stack
    -- Executing [[email protected]:13] ExecIf("SIP/1001-00000020", "0?Set(GROUP(concurrency_limit)=1001)") in new stack
    -- Executing [[email protected]:14] ExecIf("SIP/1001-00000020", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [[email protected]:15] GotoIf("SIP/1001-00000020", "0?continue") in new stack
    -- Executing [[email protected]:16] ExecIf("SIP/1001-00000020", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [[email protected]:17] Set("SIP/1001-00000020", "__TTL=64") in new stack
    -- Executing [[email protected]:18] GotoIf("SIP/1001-00000020", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [[email protected]:29] Set("SIP/1001-00000020", "CALLERID(number)=1001") in new stack
    -- Executing [[email protected]:30] Set("SIP/1001-00000020", "CALLERID(name)=JOHN DOE") in new stack
    -- Executing [[email protected]:31] GotoIf("SIP/1001-00000020", "0?cnum") in new stack
    -- Executing [[email protected]:32] Set("SIP/1001-00000020", "CDR(cnam)=JOHN DOE") in new stack
    -- Executing [[email protected]:33] Set("SIP/1001-00000020", "CDR(cnum)=1001") in new stack
    -- Executing [[email protected]:34] Set("SIP/1001-00000020", "CHANNEL(language)=en") in new stack
    -- Executing [[email protected]:2] Set("SIP/1001-00000020", "RingGroupMethod=none") in new stack
    -- Executing [[email protected]:3] Set("SIP/1001-00000020", "__EXTTOCALL=1013") in new stack
    -- Executing [[email protected]:4] Set("SIP/1001-00000020", "__PICKUPMARK=1013") in new stack
    -- Executing [[email protected]:5] Set("SIP/1001-00000020", "RT=") in new stack
    -- Executing [[email protected]:6] ExecIf("SIP/1001-00000020", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
    -- Executing [[email protected]:7] ExecIf("SIP/1001-00000020", "0?MacroExit()") in new stack
    -- Executing [[email protected]:8] Gosub("SIP/1001-00000020", "sub-record-check,s,1(exten,1013,dontcare)") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/1001-00000020", "0?initialized") in new stack
    -- Executing [[email protected]:2] Set("SIP/1001-00000020", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [[email protected]:3] Set("SIP/1001-00000020", "NOW=1480737007") in new stack
    -- Executing [[email protected]:4] Set("SIP/1001-00000020", "__DAY=02") in new stack
    -- Executing [[email protected]:5] Set("SIP/1001-00000020", "__MONTH=12") in new stack
    -- Executing [[email protected]:6] Set("SIP/1001-00000020", "__YEAR=2016") in new stack
    -- Executing [[email protected]:7] Set("SIP/1001-00000020", "__TIMESTR=20161202-225007") in new stack
    -- Executing [[email protected]:8] Set("SIP/1001-00000020", "__FROMEXTEN=1001") in new stack
    -- Executing [[email protected]:9] Set("SIP/1001-00000020", "__MON_FMT=wav") in new stack
    -- Executing [[email protected]:10] NoOp("SIP/1001-00000020", "Recordings initialized") in new stack
    -- Executing [[email protected]:11] ExecIf("SIP/1001-00000020", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [[email protected]:12] Set("SIP/1001-00000020", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [[email protected]:13] ExecIf("SIP/1001-00000020", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [[email protected]:14] GotoIf("SIP/1001-00000020", "5?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [[email protected]:17] GotoIf("SIP/1001-00000020", "1?sub-record-check,exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [[email protected]:1] NoOp("SIP/1001-00000020", "Exten Recording Check between 1001 and 1013") in new stack
    -- Executing [[email protected]:2] Set("SIP/1001-00000020", "CALLTYPE=internal") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/1001-00000020", "0?Set(CALLTYPE=)") in new stack
    -- Executing [[email protected]:4] Set("SIP/1001-00000020", "CALLEE=dontcare") in new stack
    -- Executing [[email protected]:5] ExecIf("SIP/1001-00000020", "0?Set(CALLEE=dontcare)") in new stack
    -- Executing [[email protected]:6] GotoIf("SIP/1001-00000020", "0?callee") in new stack
    -- Executing [[email protected]:7] GotoIf("SIP/1001-00000020", "1?caller") in new stack
    -- Goto (sub-record-check,exten,13)
    -- Executing [[email protected]:13] Set("SIP/1001-00000020", "RECMODE=force") in new stack
    -- Executing [[email protected]:14] ExecIf("SIP/1001-00000020", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [[email protected]:15] ExecIf("SIP/1001-00000020", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [[email protected]:16] Gosub("SIP/1001-00000020", "recordcheck,1(force,internal,1013)") in new stack
    -- Executing [[email protected]:1] NoOp("SIP/1001-00000020", "Starting recording check against force") in new stack
    -- Executing [[email protected]:2] Goto("SIP/1001-00000020", "force") in new stack
    -- Goto (sub-record-check,recordcheck,5)
    -- Executing [[email protected]:5] Set("SIP/1001-00000020", "__REC_POLICY_MODE=FORCE") in new stack
    -- Executing [[email protected]:6] GotoIf("SIP/1001-00000020", "1?startrec") in new stack
    -- Goto (sub-record-check,recordcheck,16)
    -- Executing [[email protected]:16] NoOp("SIP/1001-00000020", "Starting recording: internal, 1013") in new stack
    -- Executing [[email protected]:17] Set("SIP/1001-00000020", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
    -- Executing [[email protected]:18] Set("SIP/1001-00000020", "__CALLFILENAME=internal-1013-1001-20161202-225007-1480737007.35") in new stack
    -- Executing [[email protected]:19] MixMonitor("SIP/1001-00000020", "2016/12/02/internal-1013-1001-20161202-225007-1480737007.35.wav,ai(LOCAL_MIXMON_ID),") in new stack
    -- Executing [[email protected]:20] Set("SIP/1001-00000020", "__MIXMON_ID=0x7ff6f0071410") in new stack
    -- Executing [[email protected]:21] Set("SIP/1001-00000020", "__RECORD_ID=SIP/1001-00000020") in new stack
    -- Executing [[email protected]:22] Set("SIP/1001-00000020", "__REC_STATUS=RECORDING") in new stack
    -- Executing [[email protected]:23] Set("SIP/1001-00000020", "CDR(recordingfile)=internal-1013-1001-20161202-225007-1480737007.35.wav") in new stack
    -- Executing [[email protected]:24] Return("SIP/1001-00000020", "") in new stack
    -- Executing [[email protected]:17] Return("SIP/1001-00000020", "") in new stack
    -- Executing [[email protected]:9] GotoIf("SIP/1001-00000020", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,15)
    -- Executing [[email protected]:15] GosubIf("SIP/1001-00000020", "0?clrheader,1()") in new stack
    -- Executing [[email protected]:16] Macro("SIP/1001-00000020", "dial-one,,Ttr,1013") in new stack
    -- Executing [[email protected]:1] Set("SIP/1001-00000020", "DEXTEN=1013") in new stack
    -- Executing [[email protected]:2] ExecIf("SIP/1001-00000020", "0?Set(EXTTOCALL=1013)") in new stack
    -- Executing [[email protected]:3] Set("SIP/1001-00000020", "DIALSTATUS_CW=") in new stack
    -- Executing [[email protected]:4] GosubIf("SIP/1001-00000020", "0?screen,1()") in new stack
    -- Executing [[email protected]:5] GosubIf("SIP/1001-00000020", "0?cf,1()") in new stack
    -- Executing [[email protected]:6] GotoIf("SIP/1001-00000020", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,9)
    -- Executing [[email protected]:9] GotoIf("SIP/1001-00000020", "0?nodial") in new stack
    -- Executing [[email protected]:10] GotoIf("SIP/1001-00000020", "0?continue") in new stack
    -- Executing [[email protected]:11] Set("SIP/1001-00000020", "EXTHASCW=ENABLED") in new stack
    -- Executing [[email protected]:12] GotoIf("SIP/1001-00000020", "0?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,24)
    -- Executing [[email protected]:24] GotoIf("SIP/1001-00000020", "0?next3:continue") in new stack
    -- Goto (macro-dial-one,s,26)
    -- Executing [[email protected]:26] GotoIf("SIP/1001-00000020", "0?nodial") in new stack
    -- Executing [[email protected]:27] GosubIf("SIP/1001-00000020", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [[email protected]:1] Set("SIP/1001-00000020", "DSTRING=") in new stack
    -- Executing [[email protected]:2] Set("SIP/1001-00000020", "DEVICES=1013") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/1001-00000020", "0?Return()") in new stack
    -- Executing [[email protected]:4] ExecIf("SIP/1001-00000020", "0?Set(DEVICES=013)") in new stack
    -- Executing [[email protected]:5] Set("SIP/1001-00000020", "LOOPCNT=1") in new stack
    -- Executing [[email protected]:6] Set("SIP/1001-00000020", "ITER=1") in new stack
    -- Executing [[email protected]:7] Set("SIP/1001-00000020", "THISDIAL=SIP/1013") in new stack
    -- Executing [[email protected]:8] GosubIf("SIP/1001-00000020", "1?zap2dahdi,1()") in new stack
  == Begin MixMonitor Recording SIP/1001-00000020
    -- Executing [[email protected]:1] ExecIf("SIP/1001-00000020", "0?Return()") in new stack
    -- Executing [[email protected]:2] Set("SIP/1001-00000020", "NEWDIAL=") in new stack
    -- Executing [[email protected]:3] Set("SIP/1001-00000020", "LOOPCNT2=1") in new stack
    -- Executing [[email protected]:4] Set("SIP/1001-00000020", "ITER2=1") in new stack
    -- Executing [[email protected]:5] Set("SIP/1001-00000020", "THISPART2=SIP/1013") in new stack
    -- Executing [[email protected]:6] ExecIf("SIP/1001-00000020", "0?Set(THISPART2=DAHDI/1013)") in new stack
    -- Executing [[email protected]:7] Set("SIP/1001-00000020", "NEWDIAL=SIP/1013&") in new stack
    -- Executing [[email protected]:8] Set("SIP/1001-00000020", "ITER2=2") in new stack
    -- Executing [[email protected]:9] GotoIf("SIP/1001-00000020", "0?begin2") in new stack
    -- Executing [[email protected]:10] Set("SIP/1001-00000020", "THISDIAL=SIP/1013") in new stack
    -- Executing [[email protected]:11] Return("SIP/1001-00000020", "") in new stack
    -- Executing [[email protected]:9] GotoIf("SIP/1001-00000020", "1?docheck") in new stack
    -- Goto (macro-dial-one,dstring,14)
    -- Executing [[email protected]:14] GotoIf("SIP/1001-00000020", "0?skipset") in new stack
    -- Executing [[email protected]:15] Set("SIP/1001-00000020", "DSTRING=SIP/1013&") in new stack
    -- Executing [[email protected]:16] Set("SIP/1001-00000020", "ITER=2") in new stack
    -- Executing [[email protected]:17] GotoIf("SIP/1001-00000020", "0?begin") in new stack
    -- Executing [[email protected]:18] ExecIf("SIP/1001-00000020", "0?Return()") in new stack
    -- Executing [[email protected]:19] Set("SIP/1001-00000020", "DSTRING=SIP/1013") in new stack
    -- Executing [[email protected]:20] Return("SIP/1001-00000020", "") in new stack
    -- Executing [[email protected]:28] GotoIf("SIP/1001-00000020", "0?nodial") in new stack
    -- Executing [[email protected]:29] GotoIf("SIP/1001-00000020", "0?skiptrace") in new stack
    -- Executing [[email protected]:30] GosubIf("SIP/1001-00000020", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [[email protected]:1] Set("SIP/1001-00000020", "DB(CALLTRACE/1013)=1001") in new stack
    -- Executing [[email protected]:2] Return("SIP/1001-00000020", "") in new stack
    -- Executing [[email protected]:31] Set("SIP/1001-00000020", "D_OPTIONS=Ttr") in new stack
    -- Executing [[email protected]:32] NoOp("SIP/1001-00000020", "Blind Transfer: , Attended Transfer: , User: 1001, Alert Info: ") in new stack
    -- Executing [[email protected]:33] ExecIf("SIP/1001-00000020", "1?Set(ALERT_INFO=)") in new stack
    -- Executing [[email protected]:34] ExecIf("SIP/1001-00000020", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [[email protected]:35] ExecIf("SIP/1001-00000020", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [[email protected]:36] ExecIf("SIP/1001-00000020", "0?Set(ALERT_INFO=;volume=)") in new stack
    -- Executing [[email protected]:37] ExecIf("SIP/1001-00000020", "0?Set(ALERT_INFO=;volume=)") in new stack
    -- Executing [[email protected]:38] GosubIf("SIP/1001-00000020", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
    -- Executing [[email protected]:39] ExecIf("SIP/1001-00000020", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [[email protected]:40] GosubIf("SIP/1001-00000020", "0?qwait,1()") in new stack
    -- Executing [[email protected]:41] Set("SIP/1001-00000020", "__CWIGNORE=") in new stack
    -- Executing [[email protected]:42] Set("SIP/1001-00000020", "__KEEPCID=TRUE") in new stack
    -- Executing [[email protected]:43] GotoIf("SIP/1001-00000020", "0?usegoto,1") in new stack
    -- Executing [[email protected]:44] GotoIf("SIP/1001-00000020", "0?godial") in new stack
    -- Executing [[email protected]:45] Gosub("SIP/1001-00000020", "sub-presencestate-display,s,1(1013)") in new stack
    -- Executing [[email protected]:1] Goto("SIP/1001-00000020", "state-not_set,1") in new stack
    -- Goto (sub-presencestate-display,state-not_set,1)
    -- Executing [[email protected]:1] Set("SIP/1001-00000020", "PRESENCESTATE_DISPLAY=") in new stack
    -- Executing [[email protected]:2] Return("SIP/1001-00000020", "") in new stack
    -- Executing [[email protected]:46] Set("SIP/1001-00000020", "CONNECTEDLINE(name,i)=JANE DOE") in new stack
    -- Executing [[email protected]:47] Set("SIP/1001-00000020", "CONNECTEDLINE(num)=1013") in new stack
    -- Executing [[email protected]:48] Set("SIP/1001-00000020", "D_OPTIONS=TtrI") in new stack
    -- Executing [[email protected]:49] Macro("SIP/1001-00000020", "dialout-one-predial-hook,") in new stack
    -- Executing [[email protected]:1] MacroExit("SIP/1001-00000020", "") in new stack
    -- Executing [[email protected]:50] ExecIf("SIP/1001-00000020", "0?Set(D_OPTIONS=trII)") in new stack
    -- Executing [[email protected]: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 [[email protected]:1] NoOp("SIP/1013-00000021", "Applying SIP Headers to channel") in new stack
    -- Executing [[email protected]:2] Set("SIP/1013-00000021", "SIPHEADERKEYS=") in new stack
    -- Executing [[email protected]:3] While("SIP/1013-00000021", "0") in new stack
    -- Jumping to priority 7
    -- Executing [[email protected]: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:[email protected]:5061;transport=TLS>;tag=037ff50e
To: <sip:[email protected]: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:[email protected]:5061;transport=TLS>
P-Asserted-Identity: "JANE DOE" <sip:[email protected]>
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:[email protected]:5061;transport=TLS>;tag=037ff50e
To: <sip:[email protected]: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:[email protected]:5061;transport=TLS>
Content-Length: 0


<------------>
[2016-12-02 22:50:12] NOTICE[3786]: chan_sip.c:15601 sip_reregister:    -- Re-registration for  [email protected]
[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:[email protected]:5061;transport=TLS>;tag=037ff50e
To: <sip:[email protected]: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:[email protected]:5061;transport=TLS>
P-Asserted-Identity: "JANE DOE" <sip:[email protected]>
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:[email protected]: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:[email protected]:33059;transport=TLS>
To: <sip:[email protected]:5061;transport=TLS>;tag=as7be5cbbe
From: <sip:[email protected]: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:[email protected]: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:[email protected]:33059;transport=TLS>
To: <sip:[email protected]:5061;transport=TLS>;tag=as7be5cbbe
From: <sip:[email protected]: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:[email protected]: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:[email protected]:5061;transport=TLS>;tag=037ff50e
To: <sip:[email protected]: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:[email protected]: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:[email protected]:5061;transport=TLS>;tag=as7be5cbbe
From: <sip:[email protected]: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 [[email protected]:1] Macro("SIP/1001-00000020", "hangupcall,") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/1001-00000020", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [[email protected]:3] ExecIf("SIP/1001-00000020", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [[email protected]: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:[email protected]:33059;transport=TLS SIP/2.0
Via: SIP/2.0/TLS XXX.XXX.XXX.XXX:5061;branch=z9hG4bK0e48adc4;rport
Max-Forwards: 70
From: <sip:[email protected]:5061;transport=TLS>;tag=as7be5cbbe
To: <sip:[email protected]: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:[email protected]:33059;transport=TLS>
To: <sip:[email protected]:5061;transport=TLS>;tag=037ff50e
From: <sip:[email protected]: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  [email protected]
[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:[email protected]: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:[email protected]>;tag=as68d97055
To: <sip:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]: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:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>;tag=9789696e
From: "Unknown" <sip:[email protected]>;tag=as68d97055
Call-ID: [email protected]: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 '[email protected]: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  [email protected]
[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:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>
To: <sip:[email protected]:5061;transport=TLS>
From: <sip:[email protected]: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:[email protected]:5061;transport=TLS>;tag=35e23302
To: <sip:[email protected]: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:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>
To: <sip:[email protected]:5061;transport=TLS>
From: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=as6835358a
To: <sip:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]: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:[email protected]:5061;transport=TLS>;tag=35e23302
To: <sip:[email protected]: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:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>;expires=120
Date: Sat, 03 Dec 2016 03:51:35 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5061' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.74.70:33059:
NOTIFY sip:[email protected]: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:[email protected]>;tag=as5ea47839
To: <sip:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]: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:*[email protected];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:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>;tag=bdf7276a
From: "Unknown" <sip:[email protected]>;tag=as6835358a
Call-ID: [email protected]: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:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>;tag=a93b4676
From: "Unknown" <sip:[email protected]>;tag=as5ea47839
Call-ID: [email protected]:5061
CSeq: 102 NOTIFY
User-Agent: Zoiper rv2.8.15
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5061' Method: OPTIONS
Really destroying SIP dialog '[email protected]:5061' Method: NOTIFY
[2016-12-02 22:51:57] NOTICE[3786]: chan_sip.c:15601 sip_reregister:    -- Re-registration for  [email protected]
[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  [email protected]
[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:[email protected]: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:[email protected]>;tag=as525ea883
To: <sip:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]: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:[email protected]:33059;transport=TLS;rinstance=631fc073f821dcc1>;tag=dee5133b
From: "Unknown" <sip:[email protected]>;tag=as525ea883
Call-ID: [email protected]: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 '[email protected]:5061' Method: OPTIONS

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


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

It looks like everything is there. I would open a bug ticket with Asterisk on this issue.