[SOLVED] Asterisk 13 with Webrtc ... calls get dropped!

dr.x · December 28, 2016, 8:12pm

hey folks
i compiled strip 1.4.4 and had asterisk 13

when i do call from my soft phone to webrtc ex the call ring
but once i answer …it get dropped .

here is logs
care") in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/99100-00000006”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“SIP/99100-00000006”, “”) in new stack
– Executing [exten@sub-record-check:17] Return(“SIP/99100-00000006”, “”) in new stack
– Executing [s@macro-exten-vm:9] GotoIf(“SIP/99100-00000006”, “1?macrodial”) in new stack
– Goto (macro-exten-vm,s,15)
– Executing [s@macro-exten-vm:15] GosubIf(“SIP/99100-00000006”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:16] Macro(“SIP/99100-00000006”, “dial-one,Ttr,300”) in new stack
– Executing [s@macro-dial-one:1] Set(“SIP/99100-00000006”, “DEXTEN=300”) in new stack
– Executing [s@macro-dial-one:2] Set(“SIP/99100-00000006”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:3] GosubIf(“SIP/99100-00000006”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“SIP/99100-00000006”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:5] GotoIf(“SIP/99100-00000006”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,8)
– Executing [s@macro-dial-one:8] GotoIf(“SIP/99100-00000006”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:9] GotoIf(“SIP/99100-00000006”, “0?continue”) in new stack
– Executing [s@macro-dial-one:10] Set(“SIP/99100-00000006”, “EXTHASCW=ENABLED”) in new stack
– Executing [s@macro-dial-one:11] GotoIf(“SIP/99100-00000006”, “0?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,23)
– Executing [s@macro-dial-one:23] GotoIf(“SIP/99100-00000006”, “1?next3:continue”) in new stack
– Goto (macro-dial-one,s,24)
– Executing [s@macro-dial-one:24] ExecIf(“SIP/99100-00000006”, “0?Set(DIALSTATUS_CW=BUSY)”) in new stack
– Executing [s@macro-dial-one:25] GotoIf(“SIP/99100-00000006”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:26] GosubIf(“SIP/99100-00000006”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“SIP/99100-00000006”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“SIP/99100-00000006”, “DEVICES=300”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“SIP/99100-00000006”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“SIP/99100-00000006”, “0?Set(DEVICES=00)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“SIP/99100-00000006”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“SIP/99100-00000006”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“SIP/99100-00000006”, “THISDIAL=PJSIP/300”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“SIP/99100-00000006”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/99100-00000006”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“SIP/99100-00000006”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“SIP/99100-00000006”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“SIP/99100-00000006”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“SIP/99100-00000006”, “THISPART2=PJSIP/300”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“SIP/99100-00000006”, “0?Set(THISPART2=DAHDIIP/300)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/99100-00000006”, “NEWDIAL=PJSIP/300&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/99100-00000006”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/99100-00000006”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/99100-00000006”, “THISDIAL=PJSIP/300”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/99100-00000006”, “”) in new stack
– Executing [dstring@macro-dial-one:9] GotoIf(“SIP/99100-00000006”, “0?doset”) in new stack
– Executing [dstring@macro-dial-one:10] NoOp(“SIP/99100-00000006”, “Debug: Found PJSIP Destination PJSIP/300, updating with PJSIP_DIAL_CONTACTS”) in new stack
– Executing [dstring@macro-dial-one:11] Set(“SIP/99100-00000006”, “THISDIAL=PJSIP/300/sip:300@188.161.111.196:49461;rinstance=607c4b745b2becf3”) in new stack
– Executing [dstring@macro-dial-one:12] GotoIf(“SIP/99100-00000006”, “0?skipset”) in new stack
– Executing [dstring@macro-dial-one:13] Set(“SIP/99100-00000006”, “DSTRING=PJSIP/300/sip:300@188.161.111.196:49461;rinstance=607c4b745b2becf3&”) in new stack
– Executing [dstring@macro-dial-one:14] Set(“SIP/99100-00000006”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:15] GotoIf(“SIP/99100-00000006”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:16] ExecIf(“SIP/99100-00000006”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:17] Set(“SIP/99100-00000006”, “DSTRING=PJSIP/300/sip:300@188.161.111.196:49461;rinstance=607c4b745b2becf3”) in new stack
– Executing [dstring@macro-dial-one:18] Return(“SIP/99100-00000006”, “”) in new stack
– Executing [s@macro-dial-one:27] GotoIf(“SIP/99100-00000006”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“SIP/99100-00000006”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:29] GosubIf(“SIP/99100-00000006”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“SIP/99100-00000006”, “DB(CALLTRACE/300)=100”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“SIP/99100-00000006”, “”) in new stack
– Executing [s@macro-dial-one:30] Set(“SIP/99100-00000006”, “D_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dial-one:31] ExecIf(“SIP/99100-00000006”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [s@macro-dial-one:32] ExecIf(“SIP/99100-00000006”, “0?SIPAddHeader()”) in new stack
– Executing [s@macro-dial-one:33] ExecIf(“SIP/99100-00000006”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:34] GosubIf(“SIP/99100-00000006”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:35] Set(“SIP/99100-00000006”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:36] Set(“SIP/99100-00000006”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:37] GotoIf(“SIP/99100-00000006”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:38] GotoIf(“SIP/99100-00000006”, “0?godial”) in new stack
– Executing [s@macro-dial-one:39] Gosub(“SIP/99100-00000006”, “sub-presencestate-display,s,1(300)”) in new stack
– Executing [s@sub-presencestate-display:1] Goto(“SIP/99100-00000006”, “state-not_set,1”) in new stack
– Goto (sub-presencestate-display,state-not_set,1)
– Executing [state-not_set@sub-presencestate-display:1] Set(“SIP/99100-00000006”, “PRESENCESTATE_DISPLAY=”) in new stack
– Executing [state-not_set@sub-presencestate-display:2] Return(“SIP/99100-00000006”, “”) in new stack
– Executing [s@macro-dial-one:40] Set(“SIP/99100-00000006”, “CONNECTEDLINE(name,i)=300”) in new stack
– Executing [s@macro-dial-one:41] Set(“SIP/99100-00000006”, “CONNECTEDLINE(num)=300”) in new stack
– Executing [s@macro-dial-one:42] Set(“SIP/99100-00000006”, “D_OPTIONS=TtrI”) in new stack
– Executing [s@macro-dial-one:43] Macro(“SIP/99100-00000006”, “dialout-one-predial-hook,”) in new stack
– Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“SIP/99100-00000006”, “”) in new stack
– Executing [s@macro-dial-one:44] ExecIf(“SIP/99100-00000006”, “0?Set(D_OPTIONS=trII)”) in new stack
– Executing [s@macro-dial-one:45] Dial(“SIP/99100-00000006”, “PJSIP/300/sip:300@188.161.111.196:49461;rinstance=607c4b745b2becf3,TtrI”) in new stack
– Called PJSIP/300/sip:300@188.161.111.196:49461;rinstance=607c4b745b2becf3
– Connected line update to SIP/99100-00000006 prevented.
– PJSIP/300-00000005 is ringing
– PJSIP/300-00000005 answered SIP/99100-00000006
– Channel PJSIP/300-00000005 joined ‘simple_bridge’ basic-bridge <4e9b1f4d-5baf-4c2c-a795-b07153c798cd>
– Channel SIP/99100-00000006 joined ‘simple_bridge’ basic-bridge <4e9b1f4d-5baf-4c2c-a795-b07153c798cd>
[2016-12-28 15:11:11] WARNING[28418][C-00000006]: res_rtp_asterisk.c:2141 dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on ‘0x7fd4bc006ea0’
[2016-12-28 15:11:11] WARNING[28418][C-00000006]: res_rtp_asterisk.c:4120 ast_rtcp_read: RTCP Read error: Unspecified. Hanging up.
– Channel SIP/99100-00000006 left ‘simple_bridge’ basic-bridge <4e9b1f4d-5baf-4c2c-a795-b07153c798cd>
– Channel PJSIP/300-00000005 left ‘simple_bridge’ basic-bridge <4e9b1f4d-5baf-4c2c-a795-b07153c798cd>
== Spawn extension (macro-dial-one, s, 45) exited non-zero on ‘SIP/99100-00000006’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/99100-00000006’ in macro ‘exten-vm’
== Spawn extension (from-internal, 300, 2) exited non-zero on ‘SIP/99100-00000006’
– Executing [h@from-internal:1] Hangup(“SIP/99100-00000006”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/99100-00000006’
mudshare6*CLI>

so as you see the errors in the place :

2016-12-28 15:11:11] WARNING[28418][C-00000006]: res_rtp_asterisk.c:2141 dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on ‘0x7fd4bc006ea0’
[2016-12-28 15:11:11] WARNING[28418][C-00000006]: res_rtp_asterisk.c:4120 ast_rtcp_read: RTCP Read error: Unspecified. Hanging up.

not sure what the problem is !

hope to help me

cheers

navaismo · December 28, 2016, 10:49pm

You are using freepbx based on your log. You need to ask on their forums if your freepbx setup have all the dependencies in order to use Websockets.

About your error seems like the DTLS part is missconfigured I guess you are using self certificates but you need to share at leat the peer configuration about the dtls part and the websocket setup.

dr.x · December 29, 2016, 3:43am

thanks mate , but i want to ask general Q here .
do i need to add certificate o my browser ?

i added certificates on sever side , but not on my pc side /browser

do i need to compile PJproject with specific Flags ?
i compiled with
CFLAGS=’-DPJ_HAS_IPV6=1’ ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --libdir=/usr/lib64

also do i need to compile asterisk with specific options ?

i compiled it with
./configure --libdir=/usr/lib64 --with-srtp=/usr/include/srtp

and also compiled SRTP with
./configure CFLAGS=-fPIC --libdir=/usr/lib64

if you see what i did above is correct , can you confirm me plz ?

I’m very thankful to you helping .

cheers

dr.x · December 29, 2016, 3:33pm

here is a trace of call between sip phone and webrtc
[root@li666-187 ~]# asterisk -rvvvvvvvvvvvvvvv
Asterisk 13.13.1, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 13.13.1 currently running on li666-187 (pid = 28539)

<— SIP read from UDP:176.58.68.73:11360 —>
INVITE sip:300@212.71.237.187 SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—ef8abe35fd418a31;rport
Max-Forwards: 70
Contact:
To:
From: ;tag=4a9c4a6d
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 1 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.5 stamp 81140
Content-Length: 766

v=0
o=- 1483022298443464 1 IN IP4 176.58.68.73
s=X-Lite release 4.9.5 stamp 81140
c=IN IP4 176.58.68.73
t=0 0
a=ice-ufrag:637ec0
a=ice-pwd:5f1f8330727ea11d578399a42cc525fa
m=audio 12668 RTP/AVP 9 8 85 120 0 3 101
a=rtpmap:85 speex/8000
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:12669 IN IP4 176.58.68.73
a=candidate:1 1 UDP 659136 192.168.1.101 64620 typ host
a=candidate:2 1 UDP 659084 176.58.68.73 12665 typ srflx raddr 192.168.1.101 rport 64620
a=candidate:1 2 UDP 659134 192.168.1.101 64621 typ host
a=candidate:2 2 UDP 659082 176.58.68.73 12666 typ srflx raddr 192.168.1.101 rport 64621
a=ssrc:1898997464 cname:TmnRqhNBHcqR6ckS
<------------->
— (13 headers 20 lines) —
Sending to 176.58.68.73:11360 (NAT)
Sending to 176.58.68.73:11360 (NAT)
Using INVITE request as basis request - 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
Found peer ‘100’ for ‘100’ from 176.58.68.73:11360

<— Reliably Transmitting (NAT) to 176.58.68.73:11360 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—ef8abe35fd418a31;received=176.58.68.73;rport=11360
From: ;tag=4a9c4a6d
To: ;tag=as7dfa26cf
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 1 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0e96e9ea"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg’ in 12288 ms (Method: INVITE)

<— SIP read from UDP:176.58.68.73:11360 —>
ACK sip:300@212.71.237.187 SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—ef8abe35fd418a31;rport
Max-Forwards: 70
To: ;tag=as7dfa26cf
From: ;tag=4a9c4a6d
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 1 ACK
Content-Length: 0

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

<— SIP read from UDP:176.58.68.73:11360 —>
INVITE sip:300@212.71.237.187 SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;rport
Max-Forwards: 70
Contact:
To:
From: ;tag=4a9c4a6d
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 2 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.5 stamp 81140
Authorization: Digest username=“100”,realm=“asterisk”,nonce=“0e96e9ea”,uri="sip:300@212.71.237.187",response=“b58dcbc8a49dc167a2369022085d8834”,algorithm=MD5
Content-Length: 766

v=0
o=- 1483022298443464 1 IN IP4 176.58.68.73
s=X-Lite release 4.9.5 stamp 81140
c=IN IP4 176.58.68.73
t=0 0
a=ice-ufrag:637ec0
a=ice-pwd:5f1f8330727ea11d578399a42cc525fa
m=audio 12668 RTP/AVP 9 8 85 120 0 3 101
a=rtpmap:85 speex/8000
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:12669 IN IP4 176.58.68.73
a=candidate:1 1 UDP 659136 192.168.1.101 64620 typ host
a=candidate:2 1 UDP 659084 176.58.68.73 12665 typ srflx raddr 192.168.1.101 rport 64620
a=candidate:1 2 UDP 659134 192.168.1.101 64621 typ host
a=candidate:2 2 UDP 659082 176.58.68.73 12666 typ srflx raddr 192.168.1.101 rport 64621
a=ssrc:1898997464 cname:TmnRqhNBHcqR6ckS
<------------->
— (14 headers 20 lines) —
Sending to 176.58.68.73:11360 (NAT)
Using INVITE request as basis request - 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
Found peer ‘100’ for ‘100’ from 176.58.68.73:11360
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 85
Found RTP audio format 120
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format speex for ID 85
Found audio description format opus for ID 120
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|gsm|alaw|g722|speex|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 176.58.68.73:12668
Looking for 300 in from-internal (domain 212.71.237.187)
sip_route_dump: route/path hop:

<— Transmitting (NAT) to 176.58.68.73:11360 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;received=176.58.68.73;rport=11360
From: ;tag=4a9c4a6d
To:
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact:
Content-Length: 0

<------------>
– Executing [300@from-internal:1] Set(“SIP/100-0000005b”, “__RINGTIMER=15”) in new stack
– Executing [300@from-internal:2] Macro(“SIP/100-0000005b”, “exten-vm,novm,300,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“SIP/100-0000005b”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/100-0000005b”, “TOUCH_MONITOR=1483022298.623”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/100-0000005b”, “AMPUSER=100”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/100-0000005b”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/100-0000005b”, “1?Set(REALCALLERIDNUM=100)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/100-0000005b”, “AMPUSER=100”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/100-0000005b”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/100-0000005b”, “AMPUSERCIDNAME=100”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/100-0000005b”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/100-0000005b”, “AMPUSERCID=100”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/100-0000005b”, “_DIALOPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/100-0000005b”, “CALLERID(all)=“100” <100>”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/100-0000005b”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“SIP/100-0000005b”, “0?Set(GROUP(concurrency_limit)=100)”) in new stack
– Executing [s@macro-user-callerid:14] GosubIf(“SIP/100-0000005b”, “7?sub-ccss,s,1(macro-exten-vm,300)”) in new stack
– Executing [s@sub-ccss:1] ExecIf(“SIP/100-0000005b”, “0?Return()”) in new stack
– Executing [s@sub-ccss:2] Set(“SIP/100-0000005b”, “CCSS_SETUP=TRUE”) in new stack
– Executing [s@sub-ccss:3] GosubIf(“SIP/100-0000005b”, “0?monitor_config,1(macro-exten-vm,300):monitor_default,1(macro-exten-vm,300)”) in new stack
– Executing [monitor_default@sub-ccss:1] GotoIf(“SIP/100-0000005b”, “1?is_exten”) in new stack
– Goto (sub-ccss,monitor_default,4)
– Executing [monitor_default@sub-ccss:4] Set(“SIP/100-0000005b”, “CALLCOMPLETION(cc_monitor_policy)=generic”) in new stack
– Executing [monitor_default@sub-ccss:5] Set(“SIP/100-0000005b”, “CALLCOMPLETION(cc_max_monitors)=5”) in new stack
– Executing [monitor_default@sub-ccss:6] Return(“SIP/100-0000005b”, “TRUE”) in new stack
– Executing [s@sub-ccss:4] GosubIf(“SIP/100-0000005b”, “7?agent_config,1():agent_default,1()”) in new stack
– Executing [agent_config@sub-ccss:1] Set(“SIP/100-0000005b”, “CALLCOMPLETION(cc_agent_policy)=generic”) in new stack
– Executing [agent_config@sub-ccss:2] Set(“SIP/100-0000005b”, “CALLCOMPLETION(cc_offer_timer)=30”) in new stack
– Executing [agent_config@sub-ccss:3] Set(“SIP/100-0000005b”, “CALLCOMPLETION(ccbs_available_timer)=”) in new stack
– Executing [agent_config@sub-ccss:4] Set(“SIP/100-0000005b”, “CALLCOMPLETION(ccnr_available_timer)=”) in new stack
– Executing [agent_config@sub-ccss:5] Set(“SIP/100-0000005b”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
[2016-12-29 14:38:18] WARNING[31788][C-00000054]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
– Executing [agent_config@sub-ccss:6] ExecIf(“SIP/100-0000005b”, “1?Set(CALLCOMPLETION(cc_recall_timer)=)”) in new stack
– Executing [agent_config@sub-ccss:7] ExecIf(“SIP/100-0000005b”, “1?Set(CALLCOMPLETION(cc_max_agents)=)”) in new stack
– Executing [agent_config@sub-ccss:8] ExecIf(“SIP/100-0000005b”, “0?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/100_300@from-ccss-)”) in new stack
– Executing [agent_config@sub-ccss:9] Set(“SIP/100-0000005b”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
[2016-12-29 14:38:18] WARNING[31788][C-00000054]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
– Executing [agent_config@sub-ccss:10] Return(“SIP/100-0000005b”, “”) in new stack
– Executing [s@sub-ccss:5] Set(“SIP/100-0000005b”, “DB(AMPUSER/100/ccss/last_number)=300”) in new stack
– Executing [s@sub-ccss:6] Return(“SIP/100-0000005b”, “”) in new stack
– Executing [s@macro-user-callerid:15] GotoIf(“SIP/100-0000005b”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:16] Set(“SIP/100-0000005b”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“SIP/100-0000005b”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,28)
– Executing [s@macro-user-callerid:28] Set(“SIP/100-0000005b”, “CALLERID(number)=100”) in new stack
– Executing [s@macro-user-callerid:29] Set(“SIP/100-0000005b”, “CALLERID(name)=100”) in new stack
– Executing [s@macro-user-callerid:30] Set(“SIP/100-0000005b”, “CDR(cnum)=100”) in new stack
– Executing [s@macro-user-callerid:31] Set(“SIP/100-0000005b”, “CDR(cnam)=100”) in new stack
– Executing [s@macro-user-callerid:32] Set(“SIP/100-0000005b”, “CHANNEL(language)=en”) in new stack
– Executing [s@macro-exten-vm:2] Set(“SIP/100-0000005b”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“SIP/100-0000005b”, “__EXTTOCALL=300”) in new stack
– Executing [s@macro-exten-vm:4] Set(“SIP/100-0000005b”, “__PICKUPMARK=300”) in new stack
– Executing [s@macro-exten-vm:5] Set(“SIP/100-0000005b”, “RT=”) in new stack
– Executing [s@macro-exten-vm:6] ExecIf(“SIP/100-0000005b”, “0?Macro(vm,novm,DIRECTDIAL,)”) in new stack
– Executing [s@macro-exten-vm:7] ExecIf(“SIP/100-0000005b”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:8] Gosub(“SIP/100-0000005b”, “sub-record-check,s,1(exten,300,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/100-0000005b”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/100-0000005b”, “_RECSTATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/100-0000005b”, “NOW=1483022298”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/100-0000005b”, “__DAY=29”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/100-0000005b”, “__MONTH=12”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/100-0000005b”, “__YEAR=2016”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/100-0000005b”, “__TIMESTR=20161229-143818”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/100-0000005b”, “__FROMEXTEN=100”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/100-0000005b”, “_MONFMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/100-0000005b”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/100-0000005b”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/100-0000005b”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/100-0000005b”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/100-0000005b”, “5?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/100-0000005b”, “1?sub-record-check,exten,1”) in new stack
– Goto (sub-record-check,exten,1)
– Executing [exten@sub-record-check:1] NoOp(“SIP/100-0000005b”, “Exten Recording Check between 100 and 300”) in new stack
– Executing [exten@sub-record-check:2] Set(“SIP/100-0000005b”, “CALLTYPE=internal”) in new stack
– Executing [exten@sub-record-check:3] ExecIf(“SIP/100-0000005b”, “0?Set(CALLTYPE=)”) in new stack
– Executing [exten@sub-record-check:4] Set(“SIP/100-0000005b”, “CALLEE=dontcare”) in new stack
– Executing [exten@sub-record-check:5] ExecIf(“SIP/100-0000005b”, “0?Set(CALLEE=dontcare)”) in new stack
– Executing [exten@sub-record-check:6] GotoIf(“SIP/100-0000005b”, “0?callee”) in new stack
– Executing [exten@sub-record-check:7] GotoIf(“SIP/100-0000005b”, “1?caller”) in new stack
– Goto (sub-record-check,exten,13)
– Executing [exten@sub-record-check:13] Set(“SIP/100-0000005b”, “RECMODE=dontcare”) in new stack
– Executing [exten@sub-record-check:14] ExecIf(“SIP/100-0000005b”, “0?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:15] ExecIf(“SIP/100-0000005b”, “1?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:16] Gosub(“SIP/100-0000005b”, “recordcheck,1(dontcare,internal,300)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/100-0000005b”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/100-0000005b”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“SIP/100-0000005b”, “”) in new stack
– Executing [exten@sub-record-check:17] Return(“SIP/100-0000005b”, “”) in new stack
– Executing [s@macro-exten-vm:9] GotoIf(“SIP/100-0000005b”, “1?macrodial”) in new stack
– Goto (macro-exten-vm,s,15)
– Executing [s@macro-exten-vm:15] GosubIf(“SIP/100-0000005b”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:16] Macro(“SIP/100-0000005b”, “dial-one,Ttr,300”) in new stack
– Executing [s@macro-dial-one:1] Set(“SIP/100-0000005b”, “DEXTEN=300”) in new stack
– Executing [s@macro-dial-one:2] Set(“SIP/100-0000005b”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:3] GosubIf(“SIP/100-0000005b”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“SIP/100-0000005b”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:5] GotoIf(“SIP/100-0000005b”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,8)
– Executing [s@macro-dial-one:8] GotoIf(“SIP/100-0000005b”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:9] GotoIf(“SIP/100-0000005b”, “0?continue”) in new stack
– Executing [s@macro-dial-one:10] Set(“SIP/100-0000005b”, “EXTHASCW=ENABLED”) in new stack
– Executing [s@macro-dial-one:11] GotoIf(“SIP/100-0000005b”, “0?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,23)
– Executing [s@macro-dial-one:23] GotoIf(“SIP/100-0000005b”, “1?next3:continue”) in new stack
– Goto (macro-dial-one,s,24)
– Executing [s@macro-dial-one:24] ExecIf(“SIP/100-0000005b”, “0?Set(DIALSTATUS_CW=BUSY)”) in new stack
– Executing [s@macro-dial-one:25] GotoIf(“SIP/100-0000005b”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:26] GosubIf(“SIP/100-0000005b”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“SIP/100-0000005b”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“SIP/100-0000005b”, “DEVICES=300&99300”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“SIP/100-0000005b”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“SIP/100-0000005b”, “0?Set(DEVICES=00&99300)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“SIP/100-0000005b”, “LOOPCNT=2”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“SIP/100-0000005b”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“SIP/100-0000005b”, “THISDIAL=SIP/300”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“SIP/100-0000005b”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/100-0000005b”, “0?Return()”) in new stack

– Goto (macro-dial-one,dstring,13)
– Executing [dstring@macro-dial-one:13] Set(“SIP/100-0000005b”, “DSTRING=SIP/300&SIP/99300&”) in new stack
– Executing [dstring@macro-dial-one:14] Set(“SIP/100-0000005b”, “ITER=3”) in new stack
– Executing [dstring@macro-dial-one:15] GotoIf(“SIP/100-0000005b”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:16] ExecIf(“SIP/100-0000005b”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:17] Set(“SIP/100-0000005b”, “DSTRING=SIP/300&SIP/99300”) in new stack
– Executing [dstring@macro-dial-one:18] Return(“SIP/100-0000005b”, “”) in new stack
– Executing [s@macro-dial-one:27] GotoIf(“SIP/100-0000005b”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“SIP/100-0000005b”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:29] GosubIf(“SIP/100-0000005b”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“SIP/100-0000005b”, “DB(CALLTRACE/300)=100”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“SIP/100-0000005b”, “”) in new stack
– Executing [s@macro-dial-one:30] Set(“SIP/100-0000005b”, “D_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dial-one:31] ExecIf(“SIP/100-0000005b”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [s@macro-dial-one:32] ExecIf(“SIP/100-0000005b”, “0?SIPAddHeader()”) in new stack
– Executing [s@macro-dial-one:33] ExecIf(“SIP/100-0000005b”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:34] GosubIf(“SIP/100-0000005b”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:35] Set(“SIP/100-0000005b”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:36] Set(“SIP/100-0000005b”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:37] GotoIf(“SIP/100-0000005b”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:38] GotoIf(“SIP/100-0000005b”, “0?godial”) in new stack
– Executing [s@macro-dial-one:39] Gosub(“SIP/100-0000005b”, “sub-presencestate-display,s,1(300)”) in new stack
– Executing [s@sub-presencestate-display:1] Goto(“SIP/100-0000005b”, “state-not_set,1”) in new stack
– Goto (sub-presencestate-display,state-not_set,1)
– Executing [state-not_set@sub-presencestate-display:1] Set(“SIP/100-0000005b”, “PRESENCESTATE_DISPLAY=”) in new stack
– Executing [state-not_set@sub-presencestate-display:2] Return(“SIP/100-0000005b”, “”) in new stack
– Executing [s@macro-dial-one:40] Set(“SIP/100-0000005b”, “CONNECTEDLINE(name,i)=300”) in new stack
– Executing [s@macro-dial-one:41] Set(“SIP/100-0000005b”, “CONNECTEDLINE(num)=300”) in new stack
– Executing [s@macro-dial-one:42] Set(“SIP/100-0000005b”, “D_OPTIONS=TtrI”) in new stack
– Executing [s@macro-dial-one:43] Macro(“SIP/100-0000005b”, “dialout-one-predial-hook,”) in new stack
– Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“SIP/100-0000005b”, “”) in new stack
– Executing [s@macro-dial-one:44] ExecIf(“SIP/100-0000005b”, “0?Set(D_OPTIONS=trII)”) in new stack
– Executing [s@macro-dial-one:45] Dial(“SIP/100-0000005b”, “SIP/300&SIP/99300,TtrI”) in new stack
[2016-12-29 14:38:18] WARNING[31788][C-00000054]: app_dial.c:2525 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== DTLS ECDH initialized (secp256r1), faster PFS enabled
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Really destroying SIP dialog ‘48359e5a593338b07ffa02550cf32670@[2a01:7e00::f03c:91ff:fe2c:1ae0]:5061’ Method: INVITE
Audio is at 17686
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 176.58.68.73:11927:
INVITE sip:febr10jd@m1kvdi6pmhdk.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 212.71.237.187:5061;branch=z9hG4bK0fd49dfb
Max-Forwards: 70
From: “100” ;tag=as5893bf0b
To:
Contact:
Call-ID: 1b2ec219549a34055d454cd3601af774@212.71.237.187:5061
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76.4(13.13.1)
Date: Thu, 29 Dec 2016 14:38:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 488

v=0
o=root 228962848 228962848 IN IP4 212.71.237.187
s=Asterisk PBX 13.13.1
c=IN IP4 212.71.237.187
t=0 0
m=audio 17686 RTP/SAVPF 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 1C:8D:CA:19:51:CC:D8:26:DC:A3:B5:83:36:42:D4:DA:CD:61:A6:9C:CF:F6:40:11:FD:58:E4:48:E5:D7:1C:49
a=sendrecv

– Called SIP/99300
<— Transmitting (NAT) to 176.58.68.73:11360 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;received=176.58.68.73;rport=11360
From: ;tag=4a9c4a6d
To: ;tag=as0c40246b
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact:
Content-Length: 0

<------------>
– Connected line update to SIP/100-0000005b prevented.

<— SIP read from WS:176.58.68.73:11927 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS 212.71.237.187:5061;branch=z9hG4bK0fd49dfb
To:
From: “100” ;tag=as5893bf0b
Call-ID: 1b2ec219549a34055d454cd3601af774@212.71.237.187:5061
CSeq: 102 INVITE
Supported: timer,ice,outbound
Content-Length: 0

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

<— SIP read from WS:176.58.68.73:11927 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 212.71.237.187:5061;branch=z9hG4bK0fd49dfb
To: ;tag=6123l7cki0
From: “100” ;tag=as5893bf0b
Call-ID: 1b2ec219549a34055d454cd3601af774@212.71.237.187:5061
CSeq: 102 INVITE
Contact:
Supported: timer,ice,outbound
Content-Length: 0

<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop:
– SIP/99300-0000005c is ringing

<— Transmitting (NAT) to 176.58.68.73:11360 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;received=176.58.68.73;rport=11360
From: ;tag=4a9c4a6d
To: ;tag=as0c40246b
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact:
Content-Length: 0

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

<— SIP read from WS:176.58.68.73:11927 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WS 212.71.237.187:5061;branch=z9hG4bK0fd49dfb
To: ;tag=6123l7cki0
From: “100” ;tag=as5893bf0b
Call-ID: 1b2ec219549a34055d454cd3601af774@212.71.237.187:5061
CSeq: 102 INVITE
Supported: timer,ice,outbound
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to 176.58.68.73:11927:
ACK sip:febr10jd@m1kvdi6pmhdk.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 212.71.237.187:5061;branch=z9hG4bK0fd49dfb
Max-Forwards: 70
From: “100” ;tag=as5893bf0b
To: ;tag=6123l7cki0
Contact:
Call-ID: 1b2ec219549a34055d454cd3601af774@212.71.237.187:5061
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.4(13.13.1)
Content-Length: 0

Scheduling destruction of SIP dialog ‘1b2ec219549a34055d454cd3601af774@212.71.237.187:5061’ in 6464 ms (Method: INVITE)
== Everyone is busy/congested at this time (2:0/0/2)
– Executing [s@macro-dial-one:46] ExecIf(“SIP/100-0000005b”, “0?MacroExit()”) in new stack
– Executing [s@macro-dial-one:47] ExecIf(“SIP/100-0000005b”, “0?Set(DIALSTATUS=)”) in new stack
– Executing [s@macro-dial-one:48] GosubIf(“SIP/100-0000005b”, “0?s-CHANUNAVAIL,1()”) in new stack
– Executing [s@macro-dial-one:49] MacroExit(“SIP/100-0000005b”, “”) in new stack
– Executing [s@macro-exten-vm:17] Set(“SIP/100-0000005b”, “SV_DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:18] GosubIf(“SIP/100-0000005b”, “0?docfu,1()”) in new stack
– Executing [s@macro-exten-vm:19] GosubIf(“SIP/100-0000005b”, “0?docfb,1()”) in new stack
– Executing [s@macro-exten-vm:20] Set(“SIP/100-0000005b”, “DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:21] ExecIf(“SIP/100-0000005b”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:22] GotoIf(“SIP/100-0000005b”, “1?s-CHANUNAVAIL,1”) in new stack
– Goto (macro-exten-vm,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-exten-vm:1] GotoIf(“SIP/100-0000005b”, “0?exit,1”) in new stack
– Executing [s-CHANUNAVAIL@macro-exten-vm:2] PlayTones(“SIP/100-0000005b”, “congestion”) in new stack
– Executing [s-CHANUNAVAIL@macro-exten-vm:3] Congestion(“SIP/100-0000005b”, “10”) in new stack

<— Reliably Transmitting (NAT) to 176.58.68.73:11360 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;received=176.58.68.73;rport=11360
From: ;tag=4a9c4a6d
To: ;tag=as0c40246b
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0

<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on ‘SIP/100-0000005b’ in macro ‘exten-vm’
== Spawn extension (from-internal, 300, 2) exited non-zero on ‘SIP/100-0000005b’
– Executing [h@from-internal:1] Hangup(“SIP/100-0000005b”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-0000005b’
Retransmitting #1 (NAT) to 176.58.68.73:11360:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;received=176.58.68.73;rport=11360
From: ;tag=4a9c4a6d
To: ;tag=as0c40246b
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0

<— SIP read from UDP:176.58.68.73:11360 —>
ACK sip:300@212.71.237.187 SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;rport
Max-Forwards: 70
To: ;tag=as0c40246b
From: ;tag=4a9c4a6d
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg’ Method: ACK

<— SIP read from UDP:176.58.68.73:11360 —>
ACK sip:300@212.71.237.187 SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;rport
Max-Forwards: 70
To: ;tag=as0c40246b
From: ;tag=4a9c4a6d
Call-ID: 81140MWVmNTcyOWNmNGVmMTBjZTUwNWI5YjMzNDA4YzEzNDg
CSeq: 2 ACK
Content-Length: 0

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

<— SIP read from UDP:176.58.68.73:11360 —>

<------------->
li666-187CLI>
li666-187CLI>
li666-187CLI>
li666-187CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@li666-187 ~]#

=====================

navaismo · December 29, 2016, 4:25pm

Things in Vanilla asterisk are a little bit different from FreePBX thats why I suggested to ask in their forums. Yes you need certificates on both sides, maybe you need to read more before to start using the websockets.

I´ve asked for the peer configuration not for the compiled flag options, you need to provide configuration from asterisk and from websocket api, later both debugs, asterisk and browser.

Your last log show an error mostly associated with codec missconfiguration.

dr.x · December 29, 2016, 4:42pm

my friend thank you so much .

indeed I’m not sure where the source is the problem is it asterisk wrong compilation or config files . thats why i asked here .

and regarding to logs above , what do you mean with codecs issue ?

also i noted in logs above about important log which is
<— SIP read from WS:176.58.68.73:11927 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WS 212.71.237.187:5061;branch=z9hG4bK0fd49dfb

but not sure if it will help

cheers

navaismo · December 29, 2016, 4:52pm

If you need further help you need to provide the things that we asked for, not just assuming things. Can’t help you anymore without the information I´ve asked for.

dr.x · December 29, 2016, 5:11pm

sorry ,
here is peer of webrtc config [99300]
[99300]
deny=0.0.0.0/0.0.0.0
dtmfmode=rfc2833
canreinvite=no
host=dynamic
trustpid=yes
sendpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=ws
avpf=yes
force_avp=yes
icesupport=yes
encryption=yes
callgroup=
pickupgroup=
permit=0.0.0.0/0.0.0.0
dial=SIP/99300
secret=887f840990f
context=from-internal
mailbox=99300@device
callerid=300 <99300>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/default.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
dtlsrekey=0

http.conf config
[general]
enabled=yes
enablestatic=no
bindaddr=0.0.0.0
bindport=8088
prefix=
[root@li666-187 upgradescripts]#

[root@li666-187 upgradescripts]# ls -l /etc/asterisk/keys
total 28
-rwxrwxr-x 1 asterisk asterisk 229 Dec 29 14:08 ca.cfg
-rwxrwxr-x 1 asterisk asterisk 1826 Dec 29 14:08 ca.crt
-rwxrwxr-x 1 asterisk asterisk 3311 Dec 29 14:08 ca.key
-rwxrwxr-x 1 asterisk asterisk 1289 Dec 29 14:08 default.crt
-rwxrwxr-x 1 asterisk asterisk 615 Dec 29 14:08 default.csr
-rwxrwxr-x 1 asterisk asterisk 887 Dec 29 14:08 default.key
-rwxrwxr-x 1 asterisk asterisk 2176 Dec 29 14:08 default.pem
[root@li666-187 upgradescripts]#

rtp conf :
cat /etc/asterisk/rtp_custom.conf
icesupport=true
stunaddr:19302=stun.l.google.com

let me know if you need other config files

again i really appreciate your kind support

navaismo · December 29, 2016, 5:17pm

From asterisk it looks fine but missing codec configuration. Which codecs are enable for that peer?
What about from the WebRTC API, how it is configured?

And finally both logs when you make a call, from asterisk enable the sip debug to see the SDP negotation and from browser please copy all.

dr.x · December 29, 2016, 5:23pm

Which codecs are enable for that peer?
What about from the WebRTC API, how it is configured?

sorry ti say i don’t know what codecs enabled on peer .
but i can check codecs on sip , its as below
[102]
deny=0.0.0.0/0.0.0.0
secret=123456789
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=no
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=yes
force_avp=yes
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/102
permit=0.0.0.0/0.0.0.0
callerid=102 <102>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

navaismo · December 29, 2016, 5:32pm

Still no codec configuration there… I guess that is managed globally somewhere in the FreePBX files. If you can’t provide those settings please provide the complete debug of the call(again) there we can see the codecs used on both sides.

david551 · December 29, 2016, 5:48pm

Please mark the logs as unformatted text, otherwise a lot of important information is misinterpreted as HTML markup.

dr.x · December 29, 2016, 5:48pm

ok here is log

from sip 102 to webrtc 300
[root@mudshare6 ~]# asterisk -rvvvvvvvvvvvvvvvvvvv
Asterisk 13.13.1, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 13.13.1 currently running on mudshare6 (pid = 18401)
mudshare6CLI>
mudshare6CLI>
mudshare6*CLI>
Reliably Transmitting (NAT) to 185.6.17.20:14345:
OPTIONS sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217 SIP/2.0
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK058fc384;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@64.37.115.36:5061;tag=as58ac963d
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217
Contact: sip:Unknown@64.37.115.36:5061
Call-ID: 7a177166743e62a812e1d45f080808a3@64.37.115.36:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.4(13.13.1)
Date: Thu, 29 Dec 2016 17:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:185.6.17.20:14345 —>

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

<— SIP read from UDP:185.6.17.20:14345 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK058fc384;rport=5060
Contact: sip:185.6.17.20:14345
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=c7fbcd1b
From: “Unknown” sip:Unknown@64.37.115.36:5061;tag=as58ac963d
Call-ID: 7a177166743e62a812e1d45f080808a3@64.37.115.36:5061
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Supported: replaces
User-Agent: X-Lite release 4.9.5 stamp 81140
Allow-Events: talk, hold
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘7a177166743e62a812e1d45f080808a3@64.37.115.36:5061’ Method: OPTIONS

<— SIP read from UDP:185.6.17.20:14345 —>
INVITE sip:300@64.37.115.36 SIP/2.0
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—19c2883a8b174d4c;rport
Max-Forwards: 70
Contact: sip:102@185.6.17.20:14345
To: sip:300@64.37.115.36
From: sip:102@64.37.115.36;tag=646adc4a
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 1 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.5 stamp 81140
Content-Length: 761

v=0
o=- 1483033587774497 1 IN IP4 185.6.17.20
s=X-Lite release 4.9.5 stamp 81140
c=IN IP4 185.6.17.20
t=0 0
a=ice-ufrag:c6665f
a=ice-pwd:d069915e1391a8000c9c4e4078379df5
m=audio 14520 RTP/AVP 9 8 85 120 0 3 101
a=rtpmap:85 speex/8000
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:14521 IN IP4 185.6.17.20
a=candidate:1 1 UDP 659136 192.168.1.103 59348 typ host
a=candidate:2 1 UDP 659084 185.6.17.20 14516 typ srflx raddr 192.168.1.103 rport 59348
a=candidate:1 2 UDP 659134 192.168.1.103 59349 typ host
a=candidate:2 2 UDP 659082 185.6.17.20 14517 typ srflx raddr 192.168.1.103 rport 59349
a=ssrc:1382282678 cname:EZiOZDCr3Fc1brXQ
<------------->
— (13 headers 20 lines) —
Sending to 185.6.17.20:14345 (NAT)
Sending to 185.6.17.20:14345 (NAT)
Using INVITE request as basis request - 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
Found peer ‘102’ for ‘102’ from 185.6.17.20:14345

<— Reliably Transmitting (NAT) to 185.6.17.20:14345 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—19c2883a8b174d4c;received=185.6.17.20;rport=14345
From: sip:102@64.37.115.36;tag=646adc4a
To: sip:300@64.37.115.36;tag=as3b6aacc0
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 1 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7cba4e48"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY’ in 17344 ms (Method: INVITE)

<— SIP read from UDP:185.6.17.20:14345 —>
ACK sip:300@64.37.115.36 SIP/2.0
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—19c2883a8b174d4c;rport
Max-Forwards: 70
To: sip:300@64.37.115.36;tag=as3b6aacc0
From: sip:102@64.37.115.36;tag=646adc4a
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 1 ACK
Content-Length: 0

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

<— SIP read from UDP:185.6.17.20:14345 —>
INVITE sip:300@64.37.115.36 SIP/2.0
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—46aa2b7408192053;rport
Max-Forwards: 70
Contact: sip:102@185.6.17.20:14345
To: sip:300@64.37.115.36
From: sip:102@64.37.115.36;tag=646adc4a
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 2 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.5 stamp 81140
Authorization: Digest username=“102”,realm=“asterisk”,nonce=“7cba4e48”,uri="sip:300@64.37.115.36",response=“62d17f7478c012d98e89a61baf935910”,algorithm=MD5
Content-Length: 761

v=0
o=- 1483033587774497 1 IN IP4 185.6.17.20
s=X-Lite release 4.9.5 stamp 81140
c=IN IP4 185.6.17.20
t=0 0
a=ice-ufrag:c6665f
a=ice-pwd:d069915e1391a8000c9c4e4078379df5
m=audio 14520 RTP/AVP 9 8 85 120 0 3 101
a=rtpmap:85 speex/8000
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:14521 IN IP4 185.6.17.20
a=candidate:1 1 UDP 659136 192.168.1.103 59348 typ host
a=candidate:2 1 UDP 659084 185.6.17.20 14516 typ srflx raddr 192.168.1.103 rport 59348
a=candidate:1 2 UDP 659134 192.168.1.103 59349 typ host
a=candidate:2 2 UDP 659082 185.6.17.20 14517 typ srflx raddr 192.168.1.103 rport 59349
a=ssrc:1382282678 cname:EZiOZDCr3Fc1brXQ
<------------->
— (14 headers 20 lines) —
Sending to 185.6.17.20:14345 (NAT)
Using INVITE request as basis request - 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
Found peer ‘102’ for ‘102’ from 185.6.17.20:14345
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
[2016-12-29 12:46:28] NOTICE[18465][C-00000005]: chan_sip.c:10350 process_sdp: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 14520 RTP/AVP 9 8 85 120 0 3 101
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 85
Found RTP audio format 120
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format speex for ID 85
Found audio description format opus for ID 120
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|gsm|alaw|g722|speex|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 185.6.17.20:14520
Looking for 300 in from-internal (domain 64.37.115.36)
sip_route_dump: route/path hop: sip:102@185.6.17.20:14345

<— Transmitting (NAT) to 185.6.17.20:14345 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—46aa2b7408192053;received=185.6.17.20;rport=14345
From: sip:102@64.37.115.36;tag=646adc4a
To: sip:300@64.37.115.36
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:300@64.37.115.36:5061
Content-Length: 0

<------------>
– Executing [300@from-internal:1] Set(“SIP/102-00000006”, “__RINGTIMER=15”) in new stack
– Executing [300@from-internal:2] Macro(“SIP/102-00000006”, “exten-vm,novm,300,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“SIP/102-00000006”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/102-00000006”, “TOUCH_MONITOR=1483033588.38”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/102-00000006”, “AMPUSER=102”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/102-00000006”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/102-00000006”, “1?Set(REALCALLERIDNUM=102)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/102-00000006”, “AMPUSER=102”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/102-00000006”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/102-00000006”, “AMPUSERCIDNAME=102”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/102-00000006”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/102-00000006”, “AMPUSERCID=102”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/102-00000006”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/102-00000006”, “CALLERID(all)=“102” <102>”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/102-00000006”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“SIP/102-00000006”, “0?Set(GROUP(concurrency_limit)=102)”) in new stack
– Executing [s@macro-user-callerid:14] GosubIf(“SIP/102-00000006”, “7?sub-ccss,s,1(macro-exten-vm,300)”) in new stack
– Executing [s@sub-ccss:1] ExecIf(“SIP/102-00000006”, “0?Return()”) in new stack
– Executing [s@sub-ccss:2] Set(“SIP/102-00000006”, “CCSS_SETUP=TRUE”) in new stack
– Executing [s@sub-ccss:3] GosubIf(“SIP/102-00000006”, “0?monitor_config,1(macro-exten-vm,300):monitor_default,1(macro-exten-vm,300)”) in new stack
– Executing [monitor_default@sub-ccss:1] GotoIf(“SIP/102-00000006”, “1?is_exten”) in new stack
– Goto (sub-ccss,monitor_default,4)
– Executing [monitor_default@sub-ccss:4] Set(“SIP/102-00000006”, “CALLCOMPLETION(cc_monitor_policy)=generic”) in new stack
– Executing [monitor_default@sub-ccss:5] Set(“SIP/102-00000006”, “CALLCOMPLETION(cc_max_monitors)=5”) in new stack
– Executing [monitor_default@sub-ccss:6] Return(“SIP/102-00000006”, “TRUE”) in new stack
– Executing [s@sub-ccss:4] GosubIf(“SIP/102-00000006”, “7?agent_config,1():agent_default,1()”) in new stack
– Executing [agent_config@sub-ccss:1] Set(“SIP/102-00000006”, “CALLCOMPLETION(cc_agent_policy)=generic”) in new stack
– Executing [agent_config@sub-ccss:2] Set(“SIP/102-00000006”, “CALLCOMPLETION(cc_offer_timer)=30”) in new stack
– Executing [agent_config@sub-ccss:3] Set(“SIP/102-00000006”, “CALLCOMPLETION(ccbs_available_timer)=”) in new stack
– Executing [agent_config@sub-ccss:4] Set(“SIP/102-00000006”, “CALLCOMPLETION(ccnr_available_timer)=”) in new stack
– Executing [agent_config@sub-ccss:5] Set(“SIP/102-00000006”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
[2016-12-29 12:46:28] WARNING[18966][C-00000005]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
– Executing [agent_config@sub-ccss:6] ExecIf(“SIP/102-00000006”, “1?Set(CALLCOMPLETION(cc_recall_timer)=)”) in new stack
– Executing [agent_config@sub-ccss:7] ExecIf(“SIP/102-00000006”, “1?Set(CALLCOMPLETION(cc_max_agents)=)”) in new stack
– Executing [agent_config@sub-ccss:8] ExecIf(“SIP/102-00000006”, “0?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/102_300@from-ccss-)”) in new stack
– Executing [agent_config@sub-ccss:9] Set(“SIP/102-00000006”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
[2016-12-29 12:46:28] WARNING[18966][C-00000005]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
– Executing [agent_config@sub-ccss:10] Return(“SIP/102-00000006”, “”) in new stack
– Executing [s@sub-ccss:5] Set(“SIP/102-00000006”, “DB(AMPUSER/102/ccss/last_number)=300”) in new stack

-- Executing [exten@sub-record-check:7] GotoIf("SIP/102-00000006", "1?caller") in new stack
-- Goto (sub-record-check,exten,13)
-- Executing [exten@sub-record-check:13] Set("SIP/102-00000006", "RECMODE=dontcare") in new stack
-- Executing [exten@sub-record-check:14] ExecIf("SIP/102-00000006", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:15] ExecIf("SIP/102-00000006", "1?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:16] Gosub("SIP/102-00000006", "recordcheck,1(dontcare,internal,300)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/102-00000006", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/102-00000006", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/102-00000006", "") in new stack
-- Executing [exten@sub-record-check:17] Return("SIP/102-00000006", "") in new stack
-- Executing [s@macro-exten-vm:9] GotoIf("SIP/102-00000006", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,15)
-- Executing [s@macro-exten-vm:15] GosubIf("SIP/102-00000006", "0?clrheader,1()") in new stack
-- Executing [s@macro-exten-vm:16] Macro("SIP/102-00000006", "dial-one,,Ttr,300") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/102-00000006", "DEXTEN=300") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/102-00000006", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/102-00000006", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/102-00000006", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/102-00000006", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/102-00000006", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/102-00000006", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/102-00000006", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/102-00000006", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,23)
-- Executing [s@macro-dial-one:23] GotoIf("SIP/102-00000006", "1?next3:continue") in new stack
-- Goto (macro-dial-one,s,24)
-- Executing [s@macro-dial-one:24] ExecIf("SIP/102-00000006", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
-- Executing [s@macro-dial-one:25] GotoIf("SIP/102-00000006", "0?nodial") in new stack
-- Executing [s@macro-dial-one:26] GosubIf("SIP/102-00000006", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/102-00000006", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/102-00000006", "DEVICES=300&99300") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/102-00000006", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/102-00000006", "0?Set(DEVICES=00&99300)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/102-00000006", "LOOPCNT=2") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/102-00000006", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/102-00000006", "THISDIAL=PJSIP/300") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/102-00000006", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/102-00000006", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/102-00000006", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/102-00000006", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/102-00000006", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/102-00000006", "THISPART2=PJSIP/300") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/102-00000006", "0?Set(THISPART2=DAHDIIP/300)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/102-00000006", "NEWDIAL=PJSIP/300&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/102-00000006", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/102-00000006", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/102-00000006", "THISDIAL=PJSIP/300") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/102-00000006", "") in new stack
-- Executing [dstring@macro-dial-one:9] GotoIf("SIP/102-00000006", "0?doset") in new stack
-- Executing [dstring@macro-dial-one:10] NoOp("SIP/102-00000006", "Debug: Found PJSIP Destination PJSIP/300, updating with PJSIP_DIAL_CONTACTS") in new stack
-- Executing [dstring@macro-dial-one:11] Set("SIP/102-00000006", "THISDIAL=") in new stack
-- Executing [dstring@macro-dial-one:12] GotoIf("SIP/102-00000006", "1?skipset") in new stack
-- Goto (macro-dial-one,dstring,14)
-- Executing [dstring@macro-dial-one:14] Set("SIP/102-00000006", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:15] GotoIf("SIP/102-00000006", "1?begin") in new stack
-- Goto (macro-dial-one,dstring,7)
-- Executing [dstring@macro-dial-one:7] Set("SIP/102-00000006", "THISDIAL=SIP/99300") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/102-00000006", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/102-00000006", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/102-00000006", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/102-00000006", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/102-00000006", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/102-00000006", "THISPART2=SIP/99300") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/102-00000006", "0?Set(THISPART2=DAHDI/99300)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/102-00000006", "NEWDIAL=SIP/99300&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/102-00000006", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/102-00000006", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/102-00000006", "THISDIAL=SIP/99300") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/102-00000006", "") in new stack
-- Executing [dstring@macro-dial-one:9] GotoIf("SIP/102-00000006", "1?doset") in new stack
-- Goto (macro-dial-one,dstring,13)
-- Executing [dstring@macro-dial-one:13] Set("SIP/102-00000006", "DSTRING=SIP/99300&") in new stack
-- Executing [dstring@macro-dial-one:14] Set("SIP/102-00000006", "ITER=3") in new stack
-- Executing [dstring@macro-dial-one:15] GotoIf("SIP/102-00000006", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:16] ExecIf("SIP/102-00000006", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:17] Set("SIP/102-00000006", "DSTRING=SIP/99300") in new stack
-- Executing [dstring@macro-dial-one:18] Return("SIP/102-00000006", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/102-00000006", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/102-00000006", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:29] GosubIf("SIP/102-00000006", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/102-00000006", "DB(CALLTRACE/300)=102") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/102-00000006", "") in new stack
-- Executing [s@macro-dial-one:30] Set("SIP/102-00000006", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/102-00000006", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/102-00000006", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/102-00000006", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/102-00000006", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/102-00000006", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/102-00000006", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] GotoIf("SIP/102-00000006", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:38] GotoIf("SIP/102-00000006", "0?godial") in new stack
-- Executing [s@macro-dial-one:39] Gosub("SIP/102-00000006", "sub-presencestate-display,s,1(300)") in new stack
-- Executing [s@sub-presencestate-display:1] Goto("SIP/102-00000006", "state-not_set,1") in new stack
-- Goto (sub-presencestate-display,state-not_set,1)
-- Executing [state-not_set@sub-presencestate-display:1] Set("SIP/102-00000006", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [state-not_set@sub-presencestate-display:2] Return("SIP/102-00000006", "") in new stack
-- Executing [s@macro-dial-one:40] Set("SIP/102-00000006", "CONNECTEDLINE(name,i)=300") in new stack
-- Executing [s@macro-dial-one:41] Set("SIP/102-00000006", "CONNECTEDLINE(num)=300") in new stack
-- Executing [s@macro-dial-one:42] Set("SIP/102-00000006", "D_OPTIONS=TtrI") in new stack
-- Executing [s@macro-dial-one:43] Macro("SIP/102-00000006", "dialout-one-predial-hook,") in new stack
-- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/102-00000006", "") in new stack
-- Executing [s@macro-dial-one:44] ExecIf("SIP/102-00000006", "0?Set(D_OPTIONS=trII)") in new stack
-- Executing [s@macro-dial-one:45] Dial("SIP/102-00000006", "SIP/99300,,TtrI") in new stack

== DTLS ECDH initialized (secp256r1), faster PFS enabled
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 13564
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 185.6.17.20:14323:
INVITE sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK18530670
Max-Forwards: 70
From: “102” sip:102@64.37.115.36:5061;tag=as4240aea5
To: sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws
Contact: sip:102@64.37.115.36:5061;transport=WS
Call-ID: 3b4ae305108b5c5d5e81fea64a079ae4@64.37.115.36:5061
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76.4(13.13.1)
Date: Thu, 29 Dec 2016 17:46:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 710

v=0
o=root 684453541 684453541 IN IP4 64.37.115.36
s=Asterisk PBX 13.13.1
c=IN IP4 64.37.115.36
t=0 0
m=audio 13564 RTP/SAVPF 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:60eaacca29d7cdf24c2286fd33da8fb7
a=ice-pwd:35896d0878e3e4735df6db8426d3a931
a=candidate:H40257324 1 UDP 2130706431 64.37.115.36 13564 typ host
a=candidate:H40257324 2 UDP 2130706430 64.37.115.36 13565 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 3A:49:AB:D9:33:B7:C5:5D:2A:BD:E0:F3:6D:50:6D:A1:91:60:19:05:9D:62:10:24:38:3B:77:F9:07:23:05:5C
a=sendrecv

-- Called SIP/99300

<— Transmitting (NAT) to 185.6.17.20:14345 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—46aa2b7408192053;received=185.6.17.20;rport=14345
From: sip:102@64.37.115.36;tag=646adc4a
To: sip:300@64.37.115.36;tag=as70d90a27
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:300@64.37.115.36:5061
Content-Length: 0

<------------>
– Connected line update to SIP/102-00000006 prevented.

<— SIP read from WS:185.6.17.20:14323 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK18530670
To: sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws
From: “102” sip:102@64.37.115.36:5061;tag=as4240aea5
Call-ID: 3b4ae305108b5c5d5e81fea64a079ae4@64.37.115.36:5061
CSeq: 102 INVITE
Supported: timer,ice,outbound
Content-Length: 0

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

<— SIP read from WS:185.6.17.20:14323 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK18530670
To: sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws;tag=r6fcjucsfu
From: “102” sip:102@64.37.115.36:5061;tag=as4240aea5
Call-ID: 3b4ae305108b5c5d5e81fea64a079ae4@64.37.115.36:5061
CSeq: 102 INVITE
Contact: sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws
Supported: timer,ice,outbound
Content-Length: 0

<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop: sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws
– SIP/99300-00000007 is ringing

<— Transmitting (NAT) to 185.6.17.20:14345 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—46aa2b7408192053;received=185.6.17.20;rport=14345
From: sip:102@64.37.115.36;tag=646adc4a
To: sip:300@64.37.115.36;tag=as70d90a27
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:300@64.37.115.36:5061
Content-Length: 0

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

<— SIP read from WS:185.6.17.20:14323 —>
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK18530670
To: sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws;tag=r6fcjucsfu
From: “102” sip:102@64.37.115.36:5061;tag=as4240aea5
Call-ID: 3b4ae305108b5c5d5e81fea64a079ae4@64.37.115.36:5061
CSeq: 102 INVITE
Supported: timer,ice,outbound
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to 185.6.17.20:14323:
ACK sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK18530670
Max-Forwards: 70
From: “102” sip:102@64.37.115.36:5061;tag=as4240aea5
To: sip:qn9fdr6j@uccoop2hr817.invalid;transport=ws;tag=r6fcjucsfu
Contact: sip:102@64.37.115.36:5061;transport=WS
Call-ID: 3b4ae305108b5c5d5e81fea64a079ae4@64.37.115.36:5061
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.4(13.13.1)
Content-Length: 0

-- Redirecting update to SIP/102-00000006 prevented.
-- SIP/99300-00000007 is busy

Scheduling destruction of SIP dialog ‘3b4ae305108b5c5d5e81fea64a079ae4@64.37.115.36:5061’ in 16960 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [s@macro-dial-one:46] ExecIf(“SIP/102-00000006”, “0?MacroExit()”) in new stack
– Executing [s@macro-dial-one:47] ExecIf(“SIP/102-00000006”, “0?Set(DIALSTATUS=)”) in new stack
– Executing [s@macro-dial-one:48] GosubIf(“SIP/102-00000006”, “0?s-BUSY,1()”) in new stack
– Executing [s@macro-dial-one:49] MacroExit(“SIP/102-00000006”, “”) in new stack
– Executing [s@macro-exten-vm:17] Set(“SIP/102-00000006”, “SV_DIALSTATUS=BUSY”) in new stack
– Executing [s@macro-exten-vm:18] GosubIf(“SIP/102-00000006”, “0?docfu,1()”) in new stack
– Executing [s@macro-exten-vm:19] GosubIf(“SIP/102-00000006”, “0?docfb,1()”) in new stack
– Executing [s@macro-exten-vm:20] Set(“SIP/102-00000006”, “DIALSTATUS=BUSY”) in new stack
– Executing [s@macro-exten-vm:21] ExecIf(“SIP/102-00000006”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:22] GotoIf(“SIP/102-00000006”, “1?s-BUSY,1”) in new stack
– Goto (macro-exten-vm,s-BUSY,1)
– Executing [s-BUSY@macro-exten-vm:1] GotoIf(“SIP/102-00000006”, “0?exit,1”) in new stack
– Executing [s-BUSY@macro-exten-vm:2] PlayTones(“SIP/102-00000006”, “busy”) in new stack
– Executing [s-BUSY@macro-exten-vm:3] Busy(“SIP/102-00000006”, “20”) in new stack

<— Reliably Transmitting (NAT) to 185.6.17.20:14345 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—46aa2b7408192053;received=185.6.17.20;rport=14345
From: sip:102@64.37.115.36;tag=646adc4a
To: sip:300@64.37.115.36;tag=as70d90a27
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0

<------------>
== Spawn extension (macro-exten-vm, s-BUSY, 3) exited non-zero on ‘SIP/102-00000006’ in macro ‘exten-vm’
== Spawn extension (from-internal, 300, 2) exited non-zero on ‘SIP/102-00000006’
– Executing [h@from-internal:1] Hangup(“SIP/102-00000006”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/102-00000006’
Retransmitting #1 (NAT) to 185.6.17.20:14345:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—46aa2b7408192053;received=185.6.17.20;rport=14345
From: sip:102@64.37.115.36;tag=646adc4a
To: sip:300@64.37.115.36;tag=as70d90a27
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 2 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0

<— SIP read from UDP:185.6.17.20:14345 —>
ACK sip:300@64.37.115.36 SIP/2.0
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—46aa2b7408192053;rport
Max-Forwards: 70
To: sip:300@64.37.115.36;tag=as70d90a27
From: sip:102@64.37.115.36;tag=646adc4a
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY’ Method: ACK

<— SIP read from UDP:185.6.17.20:14345 —>
ACK sip:300@64.37.115.36 SIP/2.0
Via: SIP/2.0/UDP 185.6.17.20:14345;branch=z9hG4bK-524287-1—46aa2b7408192053;rport
Max-Forwards: 70
To: sip:300@64.37.115.36;tag=as70d90a27
From: sip:102@64.37.115.36;tag=646adc4a
Call-ID: 81140MTAzNjcxYTBhZTA2N2MzMjFlNzYxZjZlYTllNzU0YjY
CSeq: 2 ACK
Content-Length: 0

dr.x · December 29, 2016, 5:56pm

here is log
from webrtc 300 dialing 102

ser-Agent: FPBX-12.0.76.4(13.13.1)
Date: Thu, 29 Dec 2016 17:49:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:185.6.17.20:14345 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK0b82d943;rport=5060
Contact: sip:185.6.17.20:14345
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=b34bcc39
From: “Unknown” sip:Unknown@64.37.115.36:5061;tag=as7c16d6a3
Call-ID: 10881d9c3ce9e62d004ca48f7ddd6167@64.37.115.36:5061
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Supported: replaces
User-Agent: X-Lite release 4.9.5 stamp 81140
Allow-Events: talk, hold
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘10881d9c3ce9e62d004ca48f7ddd6167@64.37.115.36:5061’ Method: OPTIONS

<— SIP read from WS:185.6.17.20:14694 —>
INVITE sip:102@64.37.115.36 SIP/2.0
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK1479399
Max-Forwards: 69
To: sip:102@64.37.115.36
From: sip:99300@64.37.115.36;tag=olvpeo2kil
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6820 INVITE
Contact: sip:3n25pl00@qh9nclg4m54g.invalid;transport=ws;ob
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.30
Content-Length: 1713

v=0
o=mozilla…THIS_IS_SDPARTA-50.1.0 1220611936587935970 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 7C:B2:4C:23:8E:34:60:46:28:1B:D9:7E:97:C9:76:E6:51:10:37:83:D3:4A:1D:FB:04:DF:D9:E5:C3:5D:F3:F8
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 60593 UDP/TLS/RTP/SAVPF 109 9 0 8
c=IN IP4 192.168.1.103
a=candidate:0 1 UDP 2122252543 192.168.1.103 60593 typ host
a=candidate:1 1 UDP 2122187007 172.16.168.1 61698 typ host
a=candidate:2 1 UDP 2122121471 172.16.95.1 54698 typ host
a=candidate:3 1 UDP 2122055935 192.168.46.1 55208 typ host
a=candidate:4 1 UDP 2121990399 192.168.98.1 63127 typ host
a=candidate:5 1 UDP 2121924863 172.16.119.1 49466 typ host
a=candidate:6 1 UDP 2121859327 172.16.160.1 52002 typ host
a=candidate:0 2 UDP 2122252542 192.168.1.103 59799 typ host
a=candidate:1 2 UDP 2122187006 172.16.168.1 58071 typ host
a=candidate:2 2 UDP 2122121470 172.16.95.1 61178 typ host
a=candidate:3 2 UDP 2122055934 192.168.46.1 58076 typ host
a=candidate:4 2 UDP 2121990398 192.168.98.1 50503 typ host
a=candidate:5 2 UDP 2121924862 172.16.119.1 63693 typ host
a=candidate:6 2 UDP 2121859326 172.16.160.1 50726 typ host
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=ice-pwd:a99e34e1d2078070e1431385e5fd9500
a=ice-ufrag:d09da6c1
a=mid:sdparta_0
a=msid:{1c0d7745-bfaf-5f40-8956-ad040ab861b1} {a8e5ba92-da4a-f441-86e8-30177af57908}
a=rtcp:59799 IN IP4 192.168.1.103
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:2289193020 cname:{aad94d85-2e9c-514d-9858-e2c390639722}
<------------->
— (14 headers 40 lines) —
Using INVITE request as basis request - u3d0akelcbg8qnblf495
Found peer ‘99300’ for ‘99300’ from 185.6.17.20:14694

<— Reliably Transmitting (no NAT) to 185.6.17.20:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK1479399;received=185.6.17.20
From: sip:99300@64.37.115.36;tag=olvpeo2kil
To: sip:102@64.37.115.36;tag=as202e9af3
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6820 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4b352a0f"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘u3d0akelcbg8qnblf495’ in 16064 ms (Method: INVITE)

<— SIP read from UDP:185.6.17.20:14345 —>

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

<— SIP read from WS:185.6.17.20:14694 —>
ACK sip:102@64.37.115.36 SIP/2.0
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK1479399
To: sip:102@64.37.115.36;tag=as202e9af3
From: sip:99300@64.37.115.36;tag=olvpeo2kil
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6820 ACK
Content-Length: 0

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

<— SIP read from WS:185.6.17.20:14694 —>
INVITE sip:102@64.37.115.36 SIP/2.0
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK9103721
Max-Forwards: 69
To: sip:102@64.37.115.36
From: sip:99300@64.37.115.36;tag=olvpeo2kil
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6821 INVITE
Authorization: Digest algorithm=MD5, username=“99300”, realm=“asterisk”, nonce=“4b352a0f”, uri="sip:102@64.37.115.36", response="29bc48fe6018cc9b009043f50620cc3b"
Contact: sip:3n25pl00@qh9nclg4m54g.invalid;transport=ws;ob
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.30
Content-Length: 1713

v=0
o=mozilla…THIS_IS_SDPARTA-50.1.0 1220611936587935970 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 7C:B2:4C:23:8E:34:60:46:28:1B:D9:7E:97:C9:76:E6:51:10:37:83:D3:4A:1D:FB:04:DF:D9:E5:C3:5D:F3:F8
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 60593 UDP/TLS/RTP/SAVPF 109 9 0 8
c=IN IP4 192.168.1.103
a=candidate:0 1 UDP 2122252543 192.168.1.103 60593 typ host
a=candidate:1 1 UDP 2122187007 172.16.168.1 61698 typ host
a=candidate:2 1 UDP 2122121471 172.16.95.1 54698 typ host
a=candidate:3 1 UDP 2122055935 192.168.46.1 55208 typ host
a=candidate:4 1 UDP 2121990399 192.168.98.1 63127 typ host
a=candidate:5 1 UDP 2121924863 172.16.119.1 49466 typ host
a=candidate:6 1 UDP 2121859327 172.16.160.1 52002 typ host
a=candidate:0 2 UDP 2122252542 192.168.1.103 59799 typ host
a=candidate:1 2 UDP 2122187006 172.16.168.1 58071 typ host
a=candidate:2 2 UDP 2122121470 172.16.95.1 61178 typ host
a=candidate:3 2 UDP 2122055934 192.168.46.1 58076 typ host
a=candidate:4 2 UDP 2121990398 192.168.98.1 50503 typ host
a=candidate:5 2 UDP 2121924862 172.16.119.1 63693 typ host
a=candidate:6 2 UDP 2121859326 172.16.160.1 50726 typ host
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=ice-pwd:a99e34e1d2078070e1431385e5fd9500
a=ice-ufrag:d09da6c1
a=mid:sdparta_0
a=msid:{1c0d7745-bfaf-5f40-8956-ad040ab861b1} {a8e5ba92-da4a-f441-86e8-30177af57908}
a=rtcp:59799 IN IP4 192.168.1.103
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:2289193020 cname:{aad94d85-2e9c-514d-9858-e2c390639722}
<------------->
— (15 headers 40 lines) —
Using INVITE request as basis request - u3d0akelcbg8qnblf495
Found peer ‘99300’ for ‘99300’ from 185.6.17.20:14694
== DTLS ECDH initialized (secp256r1), faster PFS enabled
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 109
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found audio description format opus for ID 109
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.103:60593
Looking for 102 in from-internal (domain 64.37.115.36)
sip_route_dump: route/path hop: sip:3n25pl00@qh9nclg4m54g.invalid;transport=ws;ob

<— Transmitting (no NAT) to 185.6.17.20:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK9103721;received=185.6.17.20
From: sip:99300@64.37.115.36;tag=olvpeo2kil
To: sip:102@64.37.115.36
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6821 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: sip:102@64.37.115.36:5061;transport=WS
Content-Length: 0

<------------>
– Executing [102@from-internal:1] Set(“SIP/99300-0000000b”, “__RINGTIMER=15”) in new stack
– Executing [102@from-internal:2] Macro(“SIP/99300-0000000b”, “exten-vm,novm,102,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“SIP/99300-0000000b”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/99300-0000000b”, “TOUCH_MONITOR=1483033769.65”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/99300-0000000b”, “AMPUSER=99300”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/99300-0000000b”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/99300-0000000b”, “1?Set(REALCALLERIDNUM=99300)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/99300-0000000b”, “AMPUSER=300”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/99300-0000000b”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/99300-0000000b”, “AMPUSERCIDNAME=300”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/99300-0000000b”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/99300-0000000b”, “AMPUSERCID=300”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/99300-0000000b”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/99300-0000000b”, “CALLERID(all)=“300” <300>”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/99300-0000000b”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“SIP/99300-0000000b”, “0?Set(GROUP(concurrency_limit)=300)”) in new stack
– Executing [s@macro-user-callerid:14] GosubIf(“SIP/99300-0000000b”, “7?sub-ccss,s,1(macro-exten-vm,102)”) in new stack
– Executing [s@sub-ccss:1] ExecIf(“SIP/99300-0000000b”, “0?Return()”) in new stack
– Executing [s@sub-ccss:2] Set(“SIP/99300-0000000b”, “CCSS_SETUP=TRUE”) in new stack
– Executing [s@sub-ccss:3] GosubIf(“SIP/99300-0000000b”, "0?monitor_config,1(macro-exten-vm,
– Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/99300-0000000b”, “NEWDIAL=SIP/99102&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/99300-0000000b”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/99300-0000000b”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/99300-0000000b”, “THISDIAL=SIP/99102”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/99300-0000000b”, “”) in new stack
– Executing [dstring@macro-dial-one:9] GotoIf(“SIP/99300-0000000b”, “1?doset”) in new stack
– Goto (macro-dial-one,dstring,13)
– Executing [dstring@macro-dial-one:13] Set(“SIP/99300-0000000b”, “DSTRING=SIP/99102&”) in new stack
– Executing [dstring@macro-dial-one:14] Set(“SIP/99300-0000000b”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:15] GotoIf(“SIP/99300-0000000b”, “1?begin”) in new stack
– Goto (macro-dial-one,dstring,7)
– Executing [dstring@macro-dial-one:7] Set(“SIP/99300-0000000b”, “THISDIAL=SIP/102”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“SIP/99300-0000000b”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/99300-0000000b”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“SIP/99300-0000000b”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“SIP/99300-0000000b”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“SIP/99300-0000000b”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“SIP/99300-0000000b”, “THISPART2=SIP/102”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“SIP/99300-0000000b”, “0?Set(THISPART2=DAHDI/102)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/99300-0000000b”, “NEWDIAL=SIP/102&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/99300-0000000b”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/99300-0000000b”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/99300-0000000b”, “THISDIAL=SIP/102”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/99300-0000000b”, “”) in new stack
– Executing [dstring@macro-dial-one:9] GotoIf(“SIP/99300-0000000b”, “1?doset”) in new stack
– Goto (macro-dial-one,dstring,13)
– Executing [dstring@macro-dial-one:13] Set(“SIP/99300-0000000b”, “DSTRING=SIP/99102&SIP/102&”) in new stack
– Executing [dstring@macro-dial-one:14] Set(“SIP/99300-0000000b”, “ITER=3”) in new stack
– Executing [dstring@macro-dial-one:15] GotoIf(“SIP/99300-0000000b”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:16] ExecIf(“SIP/99300-0000000b”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:17] Set(“SIP/99300-0000000b”, “DSTRING=SIP/99102&SIP/102”) in new stack
– Executing [dstring@macro-dial-one:18] Return(“SIP/99300-0000000b”, “”) in new stack
– Executing [s@macro-dial-one:27] GotoIf(“SIP/99300-0000000b”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“SIP/99300-0000000b”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:29] GosubIf(“SIP/99300-0000000b”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“SIP/99300-0000000b”, “DB(CALLTRACE/102)=300”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“SIP/99300-0000000b”, “”) in new stack
– Executing [s@macro-dial-one:30] Set(“SIP/99300-0000000b”, “D_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dial-one:31] ExecIf(“SIP/99300-0000000b”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [s@macro-dial-one:32] ExecIf(“SIP/99300-0000000b”, “0?SIPAddHeader()”) in new stack
– Executing [s@macro-dial-one:33] ExecIf(“SIP/99300-0000000b”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:34] GosubIf(“SIP/99300-0000000b”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:35] Set(“SIP/99300-0000000b”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:36] Set(“SIP/99300-0000000b”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:37] GotoIf(“SIP/99300-0000000b”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:38] GotoIf(“SIP/99300-0000000b”, “0?godial”) in new stack
– Executing [s@macro-dial-one:39] Gosub(“SIP/99300-0000000b”, “sub-presencestate-display,s,1(102)”) in new stack
– Executing [s@sub-presencestate-display:1] Goto(“SIP/99300-0000000b”, “state-not_set,1”) in new stack
– Goto (sub-presencestate-display,state-not_set,1)
– Executing [state-not_set@sub-presencestate-display:1] Set(“SIP/99300-0000000b”, “PRESENCESTATE_DISPLAY=”) in new stack
– Executing [state-not_set@sub-presencestate-display:2] Return(“SIP/99300-0000000b”, “”) in new stack
– Executing [s@macro-dial-one:40] Set(“SIP/99300-0000000b”, “CONNECTEDLINE(name,i)=102”) in new stack
– Executing [s@macro-dial-one:41] Set(“SIP/99300-0000000b”, “CONNECTEDLINE(num)=102”) in new stack
– Executing [s@macro-dial-one:42] Set(“SIP/99300-0000000b”, “D_OPTIONS=TtrI”) in new stack
– Executing [s@macro-dial-one:43] Macro(“SIP/99300-0000000b”, “dialout-one-predial-hook,”) in new stack
– Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“SIP/99300-0000000b”, “”) in new stack
– Executing [s@macro-dial-one:44] ExecIf(“SIP/99300-0000000b”, “0?Set(D_OPTIONS=trII)”) in new stack
– Executing [s@macro-dial-one:45] Dial(“SIP/99300-0000000b”, “SIP/99102&SIP/102,TtrI”) in new stack
[2016-12-29 12:49:29] WARNING[19185][C-00000008]: app_dial.c:2525 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Really destroying SIP dialog ‘16e835b73b8e44305f8c25251b2c64c2@64.37.115.36:5061’ Method: INVITE
Audio is at 14406
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 185.6.17.20:14345:
INVITE sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217 SIP/2.0
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK25dc7cbe;rport
Max-Forwards: 70
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217
Contact: sip:300@64.37.115.36:5061
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76.4(13.13.1)
Date: Thu, 29 Dec 2016 17:49:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 331

v=0
o=root 1661950014 1661950014 IN IP4 64.37.115.36
s=Asterisk PBX 13.13.1
c=IN IP4 64.37.115.36
t=0 0
m=audio 14406 RTP/AVPF 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Called SIP/102

<— Transmitting (no NAT) to 185.6.17.20:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK9103721;received=185.6.17.20
From: sip:99300@64.37.115.36;tag=olvpeo2kil
To: sip:102@64.37.115.36;tag=as61a0efe2
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6821 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: sip:102@64.37.115.36:5061;transport=WS
Content-Length: 0

<------------>
– Connected line update to SIP/99300-0000000b prevented.
Retransmitting #1 (NAT) to 185.6.17.20:14345:
INVITE sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217 SIP/2.0
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK25dc7cbe;rport
Max-Forwards: 70
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217
Contact: sip:300@64.37.115.36:5061
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76.4(13.13.1)
Date: Thu, 29 Dec 2016 17:49:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 331

v=0
o=root 1661950014 1661950014 IN IP4 64.37.115.36
s=Asterisk PBX 13.13.1
c=IN IP4 64.37.115.36
t=0 0
m=audio 14406 RTP/AVPF 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— SIP read from UDP:185.6.17.20:14345 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK25dc7cbe;rport=5060
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:185.6.17.20:14345 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK25dc7cbe;rport=5060
Contact: sip:102@185.6.17.20:14345
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=7ac9d77e
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 102 INVITE
User-Agent: X-Lite release 4.9.5 stamp 81140
Allow-Events: talk, hold
Content-Length: 0

<------------->
— (10 headers 0 lines) —
sip_route_dump: route/path hop: sip:102@185.6.17.20:14345
– SIP/102-0000000c is ringing

<— Transmitting (no NAT) to 185.6.17.20:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK9103721;received=185.6.17.20
From: sip:99300@64.37.115.36;tag=olvpeo2kil
To: sip:102@64.37.115.36;tag=as61a0efe2
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6821 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: sip:102@64.37.115.36:5061;transport=WS
Content-Length: 0

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

<— SIP read from UDP:185.6.17.20:14345 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK25dc7cbe;rport=5060
Contact: sip:102@185.6.17.20:14345
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=7ac9d77e
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 102 INVITE
User-Agent: X-Lite release 4.9.5 stamp 81140
Allow-Events: talk, hold
Content-Length: 0

<------------->
— (10 headers 0 lines) —
sip_route_dump: route/path hop: sip:102@185.6.17.20:14345
– SIP/102-0000000c is ringing

<— SIP read from UDP:185.6.17.20:14345 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK25dc7cbe;rport=5060
Contact: sip:102@185.6.17.20:14345
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=7ac9d77e
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 102 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.5 stamp 81140
Content-Length: 205

v=0
o=- 2770634585 3 IN IP4 185.6.17.20
s=X-Lite release 4.9.5 stamp 81140
c=IN IP4 185.6.17.20
t=0 0
m=audio 14776 RTP/AVPF 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (12 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 185.6.17.20:14776
sip_route_dump: route/path hop: sip:102@185.6.17.20:14345
Transmitting (NAT) to 185.6.17.20:14345:
ACK sip:102@185.6.17.20:14345 SIP/2.0
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK3d0a6cac;rport
Max-Forwards: 70
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=7ac9d77e
Contact: sip:300@64.37.115.36:5061
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.4(13.13.1)
Content-Length: 0

-- Connected line update to SIP/99300-0000000b prevented.
-- SIP/102-0000000c answered SIP/99300-0000000b

Audio is at 12732
Adding codec ulaw to SDP
Adding codec alaw to SDP

<— Reliably Transmitting (no NAT) to 185.6.17.20:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK9103721;received=185.6.17.20
From: sip:99300@64.37.115.36;tag=olvpeo2kil
To: sip:102@64.37.115.36;tag=as61a0efe2
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6821 INVITE
Server: FPBX-12.0.76.4(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: sip:102@64.37.115.36:5061;transport=WS
Content-Type: application/sdp
Require: timer
Content-Length: 601

v=0
o=root 2064785174 2064785174 IN IP4 64.37.115.36
s=Asterisk PBX 13.13.1
c=IN IP4 64.37.115.36
t=0 0
m=audio 12732 RTP/SAVPF 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=ice-ufrag:77f9467b17916a5913249ed4599a5df5
a=ice-pwd:510cbe775e32ceb24e0ddd9668d4e444
a=candidate:H40257324 1 UDP 2130706431 64.37.115.36 12732 typ host
a=candidate:H40257324 2 UDP 2130706430 64.37.115.36 12733 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 3A:49:AB:D9:33:B7:C5:5D:2A:BD:E0:F3:6D:50:6D:A1:91:60:19:05:9D:62:10:24:38:3B:77:F9:07:23:05:5C
a=sendrecv

<------------>
– Channel SIP/102-0000000c joined ‘simple_bridge’ basic-bridge
– Channel SIP/99300-0000000b joined ‘simple_bridge’ basic-bridge
> 0x7fd974009780 – Probation passed - setting RTP source address to 185.6.17.20:14774
> 0x7fd974009780 – Probation passed - setting RTP source address to 185.6.17.20:14774

<— SIP read from WS:185.6.17.20:14694 —>
ACK sip:102@64.37.115.36:5061;transport=ws SIP/2.0
Via: SIP/2.0/WS qh9nclg4m54g.invalid;branch=z9hG4bK2143217
Max-Forwards: 69
To: sip:102@64.37.115.36;tag=as61a0efe2
From: sip:99300@64.37.115.36;tag=olvpeo2kil
Call-ID: u3d0akelcbg8qnblf495
CSeq: 6821 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: outbound
User-Agent: JsSIP 0.6.30
Content-Length: 0

<------------->
— (11 headers 0 lines) —
[2016-12-29 12:49:32] WARNING[19185][C-00000008]: res_rtp_asterisk.c:2141 dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on ‘0x7fd96c01dce0’
[2016-12-29 12:49:32] WARNING[19185][C-00000008]: res_rtp_asterisk.c:4120 ast_rtcp_read: RTCP Read error: Unspecified. Hanging up.
– Channel SIP/99300-0000000b left ‘simple_bridge’ basic-bridge
– Channel SIP/102-0000000c left ‘simple_bridge’ basic-bridge
Scheduling destruction of SIP dialog ‘42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061’ in 17984 ms (Method: INVITE)
Reliably Transmitting (NAT) to 185.6.17.20:14345:
BYE sip:102@185.6.17.20:14345 SIP/2.0
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK54554d06;rport
Max-Forwards: 70
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=7ac9d77e
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 103 BYE
User-Agent: FPBX-12.0.76.4(13.13.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (macro-dial-one, s, 45) exited non-zero on ‘SIP/99300-0000000b’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/99300-0000000b’ in macro ‘exten-vm’
== Spawn extension (from-internal, 102, 2) exited non-zero on ‘SIP/99300-0000000b’
– Executing [h@from-internal:1] Hangup(“SIP/99300-0000000b”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/99300-0000000b’
Scheduling destruction of SIP dialog ‘u3d0akelcbg8qnblf495’ in 16064 ms (Method: ACK)
set_destination: Parsing sip:3n25pl00@qh9nclg4m54g.invalid;transport=ws;ob for address/port to send to
set_destination: URI is for WebSocket, we can’t set destination
Reliably Transmitting (no NAT) to 185.6.17.20:5060:
BYE sip:3n25pl00@qh9nclg4m54g.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK71871d93
Max-Forwards: 70
From: sip:102@64.37.115.36;tag=as61a0efe2
To: sip:99300@64.37.115.36;tag=olvpeo2kil
Call-ID: u3d0akelcbg8qnblf495
CSeq: 102 BYE
User-Agent: FPBX-12.0.76.4(13.13.1)
Proxy-Authorization: Digest username=“qn9fdr6j”, realm=“asterisk”, algorithm=MD5, uri=“sip:64.37.115.36”, nonce=“4b352a0f”, response="25e8b8f75222fe74b496bcc5736b301d"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from WS:185.6.17.20:14694 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK71871d93
To: sip:99300@64.37.115.36;tag=olvpeo2kil
From: sip:102@64.37.115.36;tag=as61a0efe2
Call-ID: u3d0akelcbg8qnblf495
CSeq: 102 BYE
Supported: outbound
Content-Length: 0

<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Retransmitting #1 (NAT) to 185.6.17.20:14345:
BYE sip:102@185.6.17.20:14345 SIP/2.0
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK54554d06;rport
Max-Forwards: 70
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=7ac9d77e
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 103 BYE
User-Agent: FPBX-12.0.76.4(13.13.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Really destroying SIP dialog ‘u3d0akelcbg8qnblf495’ Method: ACK

<— SIP read from UDP:185.6.17.20:14345 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK54554d06;rport=5060
Contact: sip:102@185.6.17.20:14345
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=7ac9d77e
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 103 BYE
User-Agent: X-Lite release 4.9.5 stamp 81140
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061’ Method: INVITE

<— SIP read from UDP:185.6.17.20:14345 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.37.115.36:5061;branch=z9hG4bK54554d06;rport=5060
Contact: sip:102@185.6.17.20:14345
To: sip:102@185.6.17.20:14345;rinstance=60b1118452a5d217;tag=7ac9d77e
From: “300” sip:300@64.37.115.36:5061;tag=as2fc857d8
Call-ID: 42474b2f61e10acc2dfa2dda75045595@64.37.115.36:5061
CSeq: 103 BYE
User-Agent: X-Lite release 4.9.5 stamp 81140
Content-Length: 0

<------------->
— (9 headers 0 lines) —
mudshare6*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@mudshare6 ~]#

dr.x · December 29, 2016, 5:57pm

excuse i cut some of logs , becuase don’t allow me more than 32000 lines.

navaismo · December 29, 2016, 6:15pm

The first call from sip to wss show an absent peer, the second call from wss to sip was answered successfully. The codecs used are:

Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|gsm|alaw|g722|speex|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)

To see why the wss peer is absent we need the logs from the browser when it receive the call. Browser logs not the asterisk log.

dr.x · December 29, 2016, 6:27pm

call from webrtc to phone 100

GET
http://64.37.115.36/ucp/ [HTTP/1.1 200 OK 851ms]
GET
http://64.37.115.36/ucp/assets/css/compiled/main/lessphp_ac5f38ce18ed13cbed05c4e3ecb47605ff5093e6.css [HTTP/1.1 200 OK 1344ms]
GET
http://64.37.115.36/ucp/assets/css/compiled/main/lessphp_2a977093aa3d5e3e467ee7cb34d3b928a156d251.css [HTTP/1.1 200 OK 635ms]
GET
http://64.37.115.36/ucp/assets/css/compiled/main/lessphp_83a6333f9eda912ca987b39400e76bfe1ab9ce7c.css [HTTP/1.1 200 OK 381ms]
GET
http://64.37.115.36/ucp/assets/css/bootstrap-select.min.css [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/css/emojione.min.css [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/css/jquery.tokenize.css [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/css/compiled/main/lessphp_3ffc8c4df8755c26ec707a1a8d0cbdabd09a395d.css [HTTP/1.1 200 OK 646ms]
GET
http://64.37.115.36/ucp/assets/css/compiled/modules/lessphp_b07239aa36987b59873d69915593b069853b94d1.css [HTTP/1.1 200 OK 1320ms]
GET
http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/js/compiled/main/jsphpg_3d985a55b9dea7cbb38727aca2279382.js [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/js/compiled/modules/jsphp_90dee6bf5596e459deee6dc99ef95f36.js [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/fonts/lato-regular-webfont.woff [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/fonts/fontawesome-webfont.woff [HTTP/1.1 200 OK 0ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 1635ms]
POST
XHR
http://64.37.115.36/ucp/ [HTTP/1.1 200 OK 826ms]
Auto Loading ContactmanagerC jsphpg_3d985a55b9dea7cbb38727aca2279382.js:1154:1492
Auto Loading VoicemailC jsphpg_3d985a55b9dea7cbb38727aca2279382.js:1154:1492
Auto Loading WebrtcC jsphpg_3d985a55b9dea7cbb38727aca2279382.js:1154:1492
Auto Loading XmppC jsphpg_3d985a55b9dea7cbb38727aca2279382.js:1154:1492
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 612ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 631ms]
GET
http://64.37.115.36/ucp/assets/images/sangoma-logo.png [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/images/noise.png [HTTP/1.1 200 OK 0ms]
GET
http://64.37.115.36/ucp/assets/fonts/lato-bold-webfont.woff [HTTP/1.1 200 OK 0ms]
GET
XHR
http://64.37.115.36/ucp/modules/Webrtc/assets/jssiplibs/jssip-0.6.30.js [HTTP/1.1 200 OK 1845ms]
Engine connecting jsphp_90dee6bf5596e459deee6dc99ef95f36.js:40:942
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/sounds/ring.mp3 [HTTP/1.1 206 Partial Content 0ms]
GET
http://64.37.115.36:8088/ws [HTTP/1.1 101 Switching Protocols 413ms]
Engine connected jsphp_90dee6bf5596e459deee6dc99ef95f36.js:40:942
Engine registered jsphp_90dee6bf5596e459deee6dc99ef95f36.js:40:942
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 668ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 654ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 645ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 551ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 1793ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 1399ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 569ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 601ms]
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/sounds/ring.mp3 [HTTP/1.1 206 Partial Content 0ms]
Engine newRTCSession jsphp_90dee6bf5596e459deee6dc99ef95f36.js:40:942
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 409ms]
Session connecting jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/images/no_user_logo.png [HTTP/1.1 200 OK 0ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 2311ms]
Session addstream jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
Session accepted jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
Session confirmed jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/sounds/ring.mp3 [HTTP/1.1 206 Partial Content 0ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 367ms]
Session ended jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/sounds/ring.mp3 [HTTP/1.1 206 Partial Content 0ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 384ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 603ms]

dr.x · December 29, 2016, 6:28pm

from phone to webrtc

Session ended jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/sounds/ring.mp3 [HTTP/1.1 206 Partial Content 0ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 384ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 603ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 622ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 715ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 729ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 1160ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 1270ms]
Engine newRTCSession jsphp_90dee6bf5596e459deee6dc99ef95f36.js:40:942
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 559ms]
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/sounds/ring.mp3 [HTTP/1.1 206 Partial Content 0ms]
Session addstream jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
Session connecting jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
Session accepted jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/sounds/ring.mp3 [HTTP/1.1 206 Partial Content 0ms]
Session confirmed jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 476ms]
Session ended jsphp_90dee6bf5596e459deee6dc99ef95f36.js:46:68
GET
http://64.37.115.36/ucp/modules/Webrtc/assets/sounds/ring.mp3 [HTTP/1.1 206 Partial Content 0ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 707ms]
POST
XHR
http://64.37.115.36/ucp/index.php [HTTP/1.1 200 OK 496ms]

navaismo · December 29, 2016, 6:36pm

Ammm…

Those are apache logs mate, I want browser log, if you are using Chrome just press CTRL+SHIFT+J and copy all the output when you receive the call. You will see the browser’s sip debug.

dr.x · December 29, 2016, 9:14pm

I’m very sorry i will provide that now

[SOLVED] Asterisk 13 with Webrtc ... calls get dropped!

here is log from webrtc 300 dialing 102

here is log
from webrtc 300 dialing 102