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] ~]#
=====================