Troubleshooting call between Webrtc & sip client why hangup immediately!

1

hey folks
i have freepbx12/asteisk 13 with rstp and webrtc module .
the 2 endpoint call ring others

but the call always get dropped immediately once we answer it .

here is logs with sip log debug enabled

[[email protected] ~]#
[[email protected] ~]# asterisk -rvvvvvvvvvvvvvvv
Asterisk 13.13.1, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer [email protected]
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—ef8abe35fd418a31;rport
Max-Forwards: 70
Contact: sip:[email protected]:11360
To: sip:[email protected]
From: sip:[email protected];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: sip:[email protected];tag=4a9c4a6d
To: sip:[email protected];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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—ef8abe35fd418a31;rport
Max-Forwards: 70
To: sip:[email protected];tag=as7dfa26cf
From: sip:[email protected];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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;rport
Max-Forwards: 70
Contact: sip:[email protected]:11360
To: sip:[email protected]
From: sip:[email protected];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:[email protected]",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: sip:[email protected]:11360

<— 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: sip:[email protected];tag=4a9c4a6d
To: sip:[email protected]
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: sip:[email protected]:5061
Content-Length: 0

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

-- Goto (macro-dial-one,dstring,13)
-- Executing [[email protected]:13] Set("SIP/100-0000005b", "DSTRING=SIP/300&SIP/99300&") in new stack
-- Executing [[email protected]:14] Set("SIP/100-0000005b", "ITER=3") in new stack
-- Executing [[email protected]:15] GotoIf("SIP/100-0000005b", "0?begin") in new stack
-- Executing [[email protected]:16] ExecIf("SIP/100-0000005b", "0?Return()") in new stack
-- Executing [[email protected]:17] Set("SIP/100-0000005b", "DSTRING=SIP/300&SIP/99300") in new stack
-- Executing [[email protected]:18] Return("SIP/100-0000005b", "") in new stack
-- Executing [[email protected]:27] GotoIf("SIP/100-0000005b", "0?nodial") in new stack
-- Executing [[email protected]e:28] GotoIf("SIP/100-0000005b", "0?skiptrace") in new stack
-- Executing [[email protected]:29] GosubIf("SIP/100-0000005b", "1?ctset,1():ctclear,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/100-0000005b", "DB(CALLTRACE/300)=100") in new stack
-- Executing [[email protected]:2] Return("SIP/100-0000005b", "") in new stack
-- Executing [[email protected]:30] Set("SIP/100-0000005b", "D_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:31] ExecIf("SIP/100-0000005b", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [[email protected]:32] ExecIf("SIP/100-0000005b", "0?SIPAddHeader()") in new stack
-- Executing [[email protected]:33] ExecIf("SIP/100-0000005b", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [[email protected]:34] GosubIf("SIP/100-0000005b", "0?qwait,1()") in new stack
-- Executing [[email protected]:35] Set("SIP/100-0000005b", "__CWIGNORE=") in new stack
-- Executing [[email protected]:36] Set("SIP/100-0000005b", "__KEEPCID=TRUE") in new stack
-- Executing [[email protected]:37] GotoIf("SIP/100-0000005b", "0?usegoto,1") in new stack
-- Executing [[email protected]:38] GotoIf("SIP/100-0000005b", "0?godial") in new stack
-- Executing [[email protected]:39] Gosub("SIP/100-0000005b", "sub-presencestate-display,s,1(300)") in new stack
-- Executing [[email protected]:1] Goto("SIP/100-0000005b", "state-not_set,1") in new stack
-- Goto (sub-presencestate-display,state-not_set,1)
-- Executing [[email protected]:1] Set("SIP/100-0000005b", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [[email protected]:2] Return("SIP/100-0000005b", "") in new stack
-- Executing [[email protected]:40] Set("SIP/100-0000005b", "CONNECTEDLINE(name,i)=300") in new stack
-- Executing [[email protected]:41] Set("SIP/100-0000005b", "CONNECTEDLINE(num)=300") in new stack
-- Executing [[email protected]:42] Set("SIP/100-0000005b", "D_OPTIONS=TtrI") in new stack
-- Executing [[email protected]:43] Macro("SIP/100-0000005b", "dialout-one-predial-hook,") in new stack
-- Executing [[email protected]:1] MacroExit("SIP/100-0000005b", "") in new stack
-- Executing [[email protected]:44] ExecIf("SIP/100-0000005b", "0?Set(D_OPTIONS=trII)") in new stack
-- Executing [[email protected]: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 ‘[email protected][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:[email protected];transport=ws SIP/2.0
Via: SIP/2.0/WS 212.71.237.187:5061;branch=z9hG4bK0fd49dfb
Max-Forwards: 70
From: “100” sip:[email protected]:5061;tag=as5893bf0b
To: sip:[email protected];transport=ws
Contact: sip:[email protected]:5061;transport=WS
Call-ID: [email protected]: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: sip:[email protected];tag=4a9c4a6d
To: sip:[email protected];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: sip:[email protected]:5061
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: sip:[email protected];transport=ws
From: “100” sip:[email protected]:5061;tag=as5893bf0b
Call-ID: [email protected]: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: sip:[email protected];transport=ws;tag=6123l7cki0
From: “100” sip:[email protected]:5061;tag=as5893bf0b
Call-ID: [email protected]:5061
CSeq: 102 INVITE
Contact: sip:[email protected];transport=ws
Supported: timer,ice,outbound
Content-Length: 0

<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop: sip:[email protected];transport=ws
– 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: sip:[email protected];tag=4a9c4a6d
To: sip:[email protected];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: sip:[email protected]:5061
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: sip:[email protected];transport=ws;tag=6123l7cki0
From: “100” sip:[email protected]:5061;tag=as5893bf0b
Call-ID: [email protected]: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:[email protected];transport=ws SIP/2.0
Via: SIP/2.0/WS 212.71.237.187:5061;branch=z9hG4bK0fd49dfb
Max-Forwards: 70
From: “100” sip:[email protected]:5061;tag=as5893bf0b
To: sip:[email protected];transport=ws;tag=6123l7cki0
Contact: sip:[email protected]:5061;transport=WS
Call-ID: [email protected]:5061
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.4(13.13.1)
Content-Length: 0

Scheduling destruction of SIP dialog ‘[email protected]:5061’ in 6464 ms (Method: INVITE)
== Everyone is busy/congested at this time (2:0/0/2)
– Executing [[email protected]:46] ExecIf(“SIP/100-0000005b”, “0?MacroExit()”) in new stack
– Executing [[email protected]:47] ExecIf(“SIP/100-0000005b”, “0?Set(DIALSTATUS=)”) in new stack
– Executing [[email protected]:48] GosubIf(“SIP/100-0000005b”, “0?s-CHANUNAVAIL,1()”) in new stack
– Executing [[email protected]:49] MacroExit(“SIP/100-0000005b”, “”) in new stack
– Executing [[email protected]:17] Set(“SIP/100-0000005b”, “SV_DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [[email protected]:18] GosubIf(“SIP/100-0000005b”, “0?docfu,1()”) in new stack
– Executing [[email protected]:19] GosubIf(“SIP/100-0000005b”, “0?docfb,1()”) in new stack
– Executing [[email protected]:20] Set(“SIP/100-0000005b”, “DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [[email protected]:21] ExecIf(“SIP/100-0000005b”, “0?MacroExit()”) in new stack
– Executing [[email protected]:22] GotoIf(“SIP/100-0000005b”, “1?s-CHANUNAVAIL,1”) in new stack
– Goto (macro-exten-vm,s-CHANUNAVAIL,1)
– Executing [[email protected]:1] GotoIf(“SIP/100-0000005b”, “0?exit,1”) in new stack
– Executing [[email protected]:2] PlayTones(“SIP/100-0000005b”, “congestion”) in new stack
– Executing [[email protected]: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: sip:[email protected];tag=4a9c4a6d
To: sip:[email protected];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 [[email protected]: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: sip:[email protected];tag=4a9c4a6d
To: sip:[email protected];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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;rport
Max-Forwards: 70
To: sip:[email protected];tag=as0c40246b
From: sip:[email protected];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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 176.58.68.73:11360;branch=z9hG4bK-524287-1—2d710a3743d3d24e;rport
Max-Forwards: 70
To: sip:[email protected];tag=as0c40246b
From: sip:[email protected];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
[[email protected] ~]#

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

2

Upgrade to FreePBX 13. WebRTC support in browsers, Asterisk, and FreePBX has been fast-changing over the past couple years. You could probably dig in and hack through this one but it is not worth it.

1 Like
3

That looks important. Not sure what it means, but it looks important.

1 Like
4

are you sure that its freepbx issue , and not an asterisk issue ?

guys i see all people on forums complain from webrtc and didn’t find any once who ever ha did working !!!

may i ask you guys if you personally had freepbx webrtc working without problems ?

i will try freepbx 13 , but you didn’t ask me which asterisk version should i sue with freepbx 13
and if there is any important flags I’m supposed to compile with

unfortunately freepbx wiki don’t cover setup and needed steps for that setup .

many thanks and let me know if you want me post more info or config files

cheers

5

Yes it works great. I am running FreePBX 13 with Asterisk 11 from Debian 8 apt-get repo. That’s not a documented configuration as far as I know but it doesn’t matter. Others here have confirmed WebRTC works with FreePBX 13 and Asterisk 13. The main things you will get from updating to FreePBX 13 (other than a current supported version) are the new WebRTC client and the SSL certificate module. In modern browsers you need to use SSL and secure web sockets (WSS) for WebRTC. If you try plain WS or plain HTTP you will be blocked, warned, threatened, or silently defeated by the browser – in short, it won’t work.

6

thank you I’m installation freepbx 13 , i will test and let you know .

but one more thing i don’t understand !
you said
"""“n modern browsers you need to use SSL and secure web sockets (WSS) for WebRTC. If you try plain WS or plain HTTP you will be blocked, warned, threatened, or silently defeated by the browser – in short, it won’t work.”"""

so how can i let my browser (chrome) to use ssl for was not plain ?

cheers

7

If you connect over https then the WebRTC client will also choose WSS.

8

thank you

but how connect over https ?

just change from http://xxxx/ to https://xxxx or you mean other thing ?

9

now i had freepbx 13 asterisk 13 but didn’t have option to test

on webrtc peer i have it in unreachable state in asterisk !!

i tested on firefox & chrome and no luck

what do you suggest ?

10

bro do you mind if you tell me what compile options did you use for libsrtp , pjsip & asterisk ?

thank you

11

See above. I didn’t compile. Here is the documentation for the WebRTC phone. It is out of date but you should still go through it and make sure you haven’t missed any steps. http://wiki.freepbx.org/display/FPG/WebRTC+Phone-UCP

12

ok i will try .

one more question .

how can i establish secure connection from my browser ?

i was unable to test from chrome

but able from firefox

can you just guide me how to use secure connection to let chrome work at least for testing ?

many thanks

13

With the certificates in place on the server, connect to https://yourpbx/ucp and make sure you don’t get any browser errors.

14

got it

thanks