[ bogdan.kecman @ 19.07.2018. 08:08 ] @
za pocetak, ja nit ovo volim da radim nit znam, samo pomazem burazeru...

u pitanju je freePBX koji sam ja (vako polupismen za voip) namestio i koji radi godinama, lepo se upgradeuje regularno etc etc ... i onda smo stigli na 99 telefona koji rade ok (svaki telefon po 4 do 10 "linija", sve linije na jednom telefonu su isti lokal, znaci imamo 99 extension-a koji rade) ...

i sad stigne nekih 10 novih telefona, ja ih dodam kao i do sada, kreiram extension, napravim 00....cfg fajl u /tftpboot sa podacima za taj telefon i login datom za novi extension.. sve kao do sada, nista specijalno T21 yealink .. telefoni sa najnovijim firmware-om (istim kao jos jedno 30-40 takvih istih telefona)... resnem telefon, on se prijavi regularno, kaze da je dobio lokal koji mu je namenjen sve full kako treba, okrenem sa tog telefona zove super ... reko sve do jaja .. ali prc, ako probas da zovnes taj extension sa drugog lokala - ne radi, da busy i prikaze gresku na telefonu koji zove.. dodas ga u ring grupu, zoves tu ring grupu, svi zvone osim njega (tj tih novih x telefona dodatih...) ... to sto je 99 ispravnih je mozda bitno, mozda nije, ja nsiam uspeo da nadjem nigde da postoji neki limit, sve sam zivo precesljao, no opet, ja sam polupismen sto se ove price tice pa .. mozda sam preskocio nesto bitno...

napravio sam i temu/tiket na freepbx-u naravno posto ga ne placam tamo me ignorisu na keca
https://community.freepbx.org/...ome-limitation-somewhere/51276

e sad .. posto nisam nepismen nego samo polupismen, ja sam probo da debagujem malo taj asterisk ... i sip show peers lepo prikaze da se extension koji sam trazio registrovo kako treba

Code:

...
446/446                   192.168.1.204                            D  No         No          A  5066     OK (23 ms)
...


ali kada probam da zovem taj extension, taj telefon, na kome na ekranu pise da je dobio taj extension, koji se lepo registrovo na taj extension, koji je se ulogovo sa tim username-om... kaze 404 %[email protected]^&$)(#@^@

Code:

<--- SIP read from UDP:192.168.1.204:5066 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK783b3a7e
From: "475" <sip:[email protected]>;tag=as5c07bd3b
To: <sip:[email protected]:5066>;tag=294278800
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T21P 34.72.0.75
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0


ovo 404 znaci, kako mi kazu drugari i kako kaze internet, da sam telefon nema pojma ko je lokal koji se trazi (njegov lokal koji sam prikaze na ekranu) ?!?!!

reko aj, neispravan telefon, nije ni prvi ni poslednji ali sad foru, uzmem ja lepo i ugasim fizicki telefon 450 koji radi, promenim cfg fajl u tftboot za ovaj telefon koji je 446 sada i ne radi da bude 450, resnem telefon, on se butne, dobije isti stari ip (.204), dobije novi lokal 450 jeli tako se ulogovo, i RADI 1/1 !?!?!? vratim ga na 446 ... ne radi, prebacim ga na 450 radi, onaj sto je bio 450 promenim config da bude 446 upalim dobije 446 nece primi poziv ?!?!?!!?

e sad vec ispi1234 .. odspavao, odgledao film, probao ponovo, odspavao, odgledao film, probao ponovo i odustajem pa pisem ovde .. mozda neko ima ideju sta da probam. ja sam na ivici nerava

hvala unapred :)
[ rajco @ 19.07.2018. 09:42 ] @
Caos,

Daj nam verziju asteriska.
Code:
asterisk -rx 'core show version'

Posle toga uradi:
Code:
asterisk -rvvvv

Pozovi ext 446 pa nam daj komplet output.
Nakon toga uključi debug na ext 446 i neke sa koje zoves pa nam opet daj komplet rezultat.
Code:
sip set debug peer 446
sip set debug peer 476

Takođe nam daj i sadržaj sip.conf od te/tih ext.
[ bogdan.kecman @ 19.07.2018. 09:55 ] @
Code:

[[email protected] ~]# asterisk -rx 'core show version'
Asterisk 13.19.1 built by mockbuild @ jenkins2.schmoozecom.net on a i686 running Linux on 2018-02-13 20:51:18 UTC
[[email protected] ~]#


sad cu da vatam debug pa saljem (imam vec debug upaljen na onom kog zovem, nisam palio na onom koji zove), hvala
[ bogdan.kecman @ 19.07.2018. 10:14 ] @
btw dal mi treba -rvvvvv ako cu da vadim iz loga a ne sa konzole?
[ bogdan.kecman @ 19.07.2018. 10:20 ] @
conference deo (ovo freepbx dodaje sam, mada nije dodao za sve extenzije)
Code:


exten => 8446,1,Macro(user-callerid,)
exten => 8446,n,ExecIf($["${DB(CONFERENCE/8446/language)}" != ""]?Set(CHANNEL(language)=${DB(CONFERENCE/8446/language)}))
exten => 8446,n,Set(MEETME_ROOMNUM=8446)
exten => 8446,n,Set(MAX_PARTICIPANTS=0)
exten => 8446,n,Set(MEETME_MUSIC=${MOHCLASS})
exten => 8446,n,ExecIf($["${DB(CONFERENCE/8446/users)}" != ""]?Set(MAX_PARTICIPANTS=${DB(CONFERENCE/8446/users)}))
exten => 8446,n,ExecIf($["${DB(CONFERENCE/8446/music)}" != "inherit" & "${DB(CONFERENCE/8446/music)}" != ""]?Set(MEETME_MUSIC=${DB(CONFERENCE/8446/music)}))
exten => 8446,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?ANSWERED)
exten => 8446,n,Answer
exten => 8446,n,Wait(1)
exten => 8446,n(ANSWERED),GotoIf($["${DB(CONFERENCE/8446/userpin)}" = "" & "${DB(CONFERENCE/8446/adminpin)}" = ""]?USER:CHECKPIN)
exten => 8446,n(CHECKPIN),GotoIf($["${PIN}" = ""]?READPIN)
exten => 8446,n,GotoIf($["${DB(CONFERENCE/8446/userpin)}" = "" & "${DB(CONFERENCE/8446/adminpin)}" = ""]?USER)
exten => 8446,n,GotoIf($["${DB(CONFERENCE/8446/userpin)}" != "" & "${PIN}" = "${DB(CONFERENCE/8446/userpin)}"]?USER)
exten => 8446,n,GotoIf($["${DB(CONFERENCE/8446/adminpin)}" != "" & "${PIN}" = "${DB(CONFERENCE/8446/adminpin)}"]?ADMIN)
exten => 8446,n(READPIN),Set(PINCOUNT=0)
exten => 8446,n(RETRYPIN),GosubIf($[${DIALPLAN_EXISTS(ext-meetme-lang-playback,${CHANNEL(language)})}]?ext-meetme-lang-playback,${CHANNEL(language)},retrypin():ext-meetme-lang-playback,en,retrypin())
exten => 8446,n,GotoIf($["${DB(CONFERENCE/8446/userpin)}" != "" & "${PIN}" = "${DB(CONFERENCE/8446/userpin)}"]?USER)
exten => 8446,n,GotoIf($["${DB(CONFERENCE/8446/adminpin)}" != "" & "${PIN}" = "${DB(CONFERENCE/8446/adminpin)}"]?ADMIN)
exten => 8446,n,GotoIf($["${DB(CONFERENCE/8446/userpin)}" = ""]?USER)
exten => 8446,n,Set(PINCOUNT=$[${PINCOUNT}+1])
exten => 8446,n,GotoIf($[${PINCOUNT}>3]?h,1)
exten => 8446,n,Playback(conf-invalidpin)
exten => 8446,n,Goto(RETRYPIN)
exten => 8446,n(ADMIN),Gosub(sub-conference-options,s,1(8446,ADMIN))
exten => 8446,n,Gosub(sub-record-check,s,1(conf,8446,never))
exten => 8446,n,ExecIf($["${DB(CONFERENCE/8446/joinmsg)}" != ""]?Playback(${DB(CONFERENCE/8446/joinmsg)}))
exten => 8446,n,Goto(STARTMEETME,1)
exten => 8446,n(USER),Gosub(sub-conference-options,s,1(8446,USER))
exten => 8446,n,Gosub(sub-record-check,s,1(conf,8446,never))
exten => 8446,n,ExecIf($["${DB(CONFERENCE/8446/joinmsg)}" != ""]?Playback(${DB(CONFERENCE/8446/joinmsg)}))
exten => 8446,n,Goto(STARTMEETME,1)
exten => 8446,hint,confbridge:8446


sta god da je ovo:
Code:

exten => 446,1,GotoIf($[${DB_EXISTS(AMPUSER/${EXTEN}/followme/ddial)} != 1 | "${DB(AMPUSER/${EXTEN}/followme/ddial)}" = "EXTENSION" ]?ext-local,${EXTEN},1:followme-check,${EXTEN},1)


Code:

exten => _RG-446.,1+1,Macro(dial,${DB(AMPUSER/446/followme/grptime)},${DIAL_OPTIONS}M(confirm^^^446),${EXTEN:7})


Code:


exten => **446,1,Macro(user-callerid,)
exten => **446,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => **446,n,Pickup(446&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected])
exten => **446,n,Hangup

exten => ***80446,1,Macro(user-callerid,)
exten => ***80446,n,Set(PICKUP_EXTEN=${AMPUSER})
exten => ***80446,n,Pickup(446&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected]&[email protected])
exten => ***80446,n,Hangup


Code:


exten => 446,1,Set(__RINGTIMER=${IF($["${DB(AMPUSER/446/ringtimer)}" > "0"]?${DB(AMPUSER/446/ringtimer)}:${RINGTIMER_DEFAULT})})
exten => 446,n,Macro(exten-vm,novm,446,0,0,0)
exten => 446,n(dest),Set(__PICKUPMARK=)
exten => 446,n,Goto(${IVR_CONTEXT},return,1)
exten => 446,hint,SIP/446&Custom:DND446,CustomPresence:446



i sip_additional.conf
Code:


[446]
deny=0.0.0.0/0.0.0.0
secret=***********
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=no
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/446
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=446 <446>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic


[ bogdan.kecman @ 19.07.2018. 10:24 ] @
Code:

ocalhost*CLI> sip set debug peer 446
SIP Debugging Enabled for IP: 192.168.1.204
localhost*CLI> sip set debug peer 473
SIP Debugging Enabled for IP: 192.168.1.129

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3280137922
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21760 21760 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12592 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.1.129:5060 (NAT)
Sending to 192.168.1.129:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3280137922;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as52961244
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48d797f9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3280137922
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as52961244
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="473", realm="asterisk", nonce="48d797f9", uri="sip:[email protected]:5060", response="c125f62eeee5e5b9dabc37ebe18b0cdf", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21760 21760 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12592 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.1.129:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.129:12592
Looking for 446 in from-internal (domain 192.168.1.10)
sip_route_dump: route/path hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 11:23:56] NOTICE[28305][C-00000000]: func_audiohookinherit.c:64 func_inheritance_write: AUDIOHOOK_INHERIT is deprecated and now does nothing.

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as140f1f52
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
[2018-07-19 11:23:56] WARNING[28305][C-00000000]: translate.c:407 framein: no samples for ulawtolin

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as140f1f52
Call-ID: [email protected]2.168.1.129
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as140f1f52
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


[ bogdan.kecman @ 19.07.2018. 10:29 ] @
ili sa malo vise detalja

Code:

[2018-07-19 11:21:34] VERBOSE[21211] asterisk.c: Asterisk Ready.
[2018-07-19 11:21:34] NOTICE[21329] chan_sip.c: Peer 'TrunkAccount2' is now Reachable. (172ms / 2000ms)
[2018-07-19 11:21:34] NOTICE[21329] chan_sip.c: Peer '502' is now Reachable. (103ms / 2000ms)
[2018-07-19 11:21:34] NOTICE[21329] chan_sip.c: Peer '501' is now Reachable. (107ms / 2000ms)
[2018-07-19 11:21:34] NOTICE[21329] chan_sip.c: Peer '504' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:34] NOTICE[21329] chan_sip.c: Peer '506' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:34] NOTICE[21329] chan_sip.c: Peer '505' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '401' is now Reachable. (86ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '403' is now Reachable. (78ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '402' is now Reachable. (79ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '405' is now Reachable. (62ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '404' is now Reachable. (86ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '407' is now Reachable. (64ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '406' is now Reachable. (72ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '409' is now Reachable. (23ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '408' is now Reachable. (73ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '412' is now Reachable. (13ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '410' is now Reachable. (19ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '413' is now Reachable. (16ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '414' is now Reachable. (67ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '415' is now Reachable. (21ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '416' is now Reachable. (68ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '418' is now Reachable. (100ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '417' is now Reachable. (113ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '419' is now Reachable. (80ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '423' is now Reachable. (82ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '421' is now Reachable. (65ms / 2000ms)
[2018-07-19 11:21:35] NOTICE[21329] chan_sip.c: Peer '422' is now Reachable. (68ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '427' is now Reachable. (85ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '420' is now Reachable. (87ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '425' is now Reachable. (71ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '426' is now Reachable. (82ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '424' is now Reachable. (78ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '429' is now Reachable. (86ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '428' is now Reachable. (81ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '432' is now Reachable. (88ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '430' is now Reachable. (90ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '433' is now Reachable. (119ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '431' is now Reachable. (71ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '434' is now Reachable. (103ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '437' is now Reachable. (105ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '435' is now Reachable. (73ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '438' is now Reachable. (75ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '439' is now Reachable. (72ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '445' is now Reachable. (84ms / 2000ms)
[2018-07-19 11:21:36] NOTICE[21329] chan_sip.c: Peer '447' is now Reachable. (70ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '444' is now Reachable. (86ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '446' is now Reachable. (20ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '441' is now Reachable. (69ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '443' is now Reachable. (74ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '440' is now Reachable. (83ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '442' is now Reachable. (70ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '449' is now Reachable. (70ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '454' is now Reachable. (68ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '456' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '455' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '457' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '450' is now Reachable. (81ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '451' is now Reachable. (59ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '452' is now Reachable. (75ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '458' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '453' is now Reachable. (73ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '459' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '467' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '465' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:37] NOTICE[21329] chan_sip.c: Peer '466' is now Reachable. (9ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '463' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '461' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '462' is now Reachable. (88ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '460' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '468' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '476' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '477' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '474' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '475' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '473' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '478' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '479' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '489' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:38] NOTICE[21329] chan_sip.c: Peer '488' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:39] NOTICE[21329] chan_sip.c: Peer '480' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:39] NOTICE[21329] chan_sip.c: Peer '481' is now Reachable. (10ms / 2000ms)
[2018-07-19 11:21:39] NOTICE[21329] chan_sip.c: Peer '483' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:39] NOTICE[21329] chan_sip.c: Peer '482' is now Reachable. (9ms / 2000ms)
[2018-07-19 11:21:39] NOTICE[21329] chan_sip.c: Peer '484' is now Reachable. (7ms / 2000ms)
[2018-07-19 11:21:39] NOTICE[21329] chan_sip.c: Peer '487' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:39] NOTICE[21329] chan_sip.c: Peer '486' is now Reachable. (8ms / 2000ms)
[2018-07-19 11:21:39] NOTICE[21329] chan_sip.c: Peer '498' is now Reachable. (10ms / 2000ms)
[2018-07-19 11:21:42] NOTICE[21329] chan_sip.c: Peer '464' is now UNREACHABLE!  Last qualify: 0
[2018-07-19 11:23:21] VERBOSE[21226] asterisk.c: Remote UNIX connection
[2018-07-19 11:23:27] VERBOSE[21329] chan_sip.c: Registered SIP '491' at 192.168.1.138:5062
[2018-07-19 11:23:29] VERBOSE[21329] chan_sip.c: Registered SIP '491' at 192.168.1.138:5064
[2018-07-19 11:23:29] VERBOSE[21329] chan_sip.c: Registered SIP '491' at 192.168.1.138:5065
[2018-07-19 11:23:32] VERBOSE[21329] chan_sip.c: Registered SIP '491' at 192.168.1.138:5066
[2018-07-19 11:23:32] VERBOSE[21329] chan_sip.c: Registered SIP '491' at 192.168.1.138:5063
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3280137922
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21760 21760 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12592 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c: --- (14 headers 15 lines) ---
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c: Sending to 192.168.1.129:5060 (NAT)
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Sending to 192.168.1.129:5060 (NAT)
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Using INVITE request as basis request - [email protected]
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found peer '473' for '473' from 192.168.1.129:5060
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3280137922;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as52961244
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48d797f9"
Content-Length: 0


<------------>
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3280137922
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as52961244
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

<------------->
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c: --- (7 headers 0 lines) ---
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="473", realm="asterisk", nonce="48d797f9", uri="sip:[email protected]:5060", response="c125f62eeee5e5b9dabc37ebe18b0cdf", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21760 21760 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12592 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c: --- (15 headers 15 lines) ---
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Sending to 192.168.1.129:5060 (no NAT)
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Using INVITE request as basis request - [email protected]
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found peer '473' for '473' from 192.168.1.129:5060
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] netsock2.c: Using SIP RTP TOS bits 184
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] netsock2.c: Using SIP RTP CoS mark 5
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found RTP audio format 0
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found RTP audio format 8
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found RTP audio format 18
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found RTP audio format 9
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found RTP audio format 101
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found audio description format G729 for ID 18
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found audio description format G722 for ID 9
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Peer audio RTP is at port 192.168.1.129:12592
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c: Looking for 446 in from-internal (domain 192.168.1.10)
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] sip/route.c: sip_route_dump: route/path hop: <sip:[email protected]:5060>
[2018-07-19 11:23:56] VERBOSE[21329][C-00000000] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]192.168.1.10:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] GotoIf("SIP/473-00000000", "1?ext-local,446,1:followme-check,446,1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (ext-local,446,1)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Set("SIP/473-00000000", "__RINGTIMER=15") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Macro("SIP/473-00000000", "exten-vm,novm,446,0,0,0") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Macro("SIP/473-00000000", "user-callerid,") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Set("SIP/473-00000000", "TOUCH_MONITOR=1531992236.0") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Set("SIP/473-00000000", "AMPUSER=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] GotoIf("SIP/473-00000000", "0?report") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] ExecIf("SIP/473-00000000", "1?Set(REALCALLERIDNUM=473)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:5] Set("SIP/473-00000000", "AMPUSER=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:6] GotoIf("SIP/473-00000000", "0?limit") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:7] Set("SIP/473-00000000", "AMPUSERCIDNAME=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:8] ExecIf("SIP/473-00000000", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:9] GotoIf("SIP/473-00000000", "0?report") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:10] Set("SIP/473-00000000", "AMPUSERCID=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:11] Set("SIP/473-00000000", "__DIAL_OPTIONS=Ttr") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]rid:12] Set("SIP/473-00000000", "CALLERID(all)="473" <473>") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:13] GotoIf("SIP/473-00000000", "0?limit") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:14] ExecIf("SIP/473-00000000", "0?Set(GROUP(concurrency_limit)=473)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:15] ExecIf("SIP/473-00000000", "0?Set(CHANNEL(language)=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:16] NoOp("SIP/473-00000000", "Macro Depth is 2") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:17] GotoIf("SIP/473-00000000", "1?report2:macroerror") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:18] GotoIf("SIP/473-00000000", "0?continue") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:19] ExecIf("SIP/473-00000000", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:20] Set("SIP/473-00000000", "__TTL=64") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:21] GotoIf("SIP/473-00000000", "1?continue") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:37] Set("SIP/473-00000000", "CALLERID(number)=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:38] Set("SIP/473-00000000", "CALLERID(name)=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:39] GotoIf("SIP/473-00000000", "0?cnum") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:40] Set("SIP/473-00000000", "CDR(cnam)=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:41] Set("SIP/473-00000000", "CDR(cnum)=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:42] Set("SIP/473-00000000", "CHANNEL(language)=en") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Set("SIP/473-00000000", "RingGroupMethod=none") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] Set("SIP/473-00000000", "__EXTTOCALL=446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] Set("SIP/473-00000000", "__PICKUPMARK=446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:5] Set("SIP/473-00000000", "RT=") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:6] ExecIf("SIP/473-00000000", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:7] ExecIf("SIP/473-00000000", "0?MacroExit()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:8] ExecIf("SIP/473-00000000", "0?Gosub(ext-intercom,*80446,1())") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:9] ExecIf("SIP/473-00000000", "0?MacroExit()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:10] ExecIf("SIP/473-00000000", "0?ChanSpy(SIP/446,q)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:11] ExecIf("SIP/473-00000000", "0?MacroExit()") in new stack
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:12] ExecIf("SIP/473-00000000", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:13] ExecIf("SIP/473-00000000", "0?MacroExit()") in new stack
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:14] ExecIf("SIP/473-00000000", "0?Gosub(ext-intercom,*80446,1())") in new stack
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:15] ExecIf("SIP/473-00000000", "0?MacroExit()") in new stack
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:16] ExecIf("SIP/473-00000000", "0?ChanSpy(SIP/446,q)") in new stack
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:17] ExecIf("SIP/473-00000000", "0?MacroExit()") in new stack
[2018-07-19 11:23:56] ERROR[28305][C-00000000] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:18] Gosub("SIP/473-00000000", "sub-record-check,s,1(exten,446,dontcare)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] GotoIf("SIP/473-00000000", "0?initialized") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Set("SIP/473-00000000", "__REC_STATUS=INITIALIZED") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] Set("SIP/473-00000000", "NOW=1531992236") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] Set("SIP/473-00000000", "__DAY=19") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:5] Set("SIP/473-00000000", "__MONTH=07") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:6] Set("SIP/473-00000000", "__YEAR=2018") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:7] Set("SIP/473-00000000", "__TIMESTR=20180719-112356") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:8] Set("SIP/473-00000000", "__FROMEXTEN=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:9] Set("SIP/473-00000000", "__MON_FMT=wav") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:10] NoOp("SIP/473-00000000", "Recordings initialized") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:11] ExecIf("SIP/473-00000000", "0?Set(ARG3=dontcare)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:12] Set("SIP/473-00000000", "REC_POLICY_MODE_SAVE=") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:13] ExecIf("SIP/473-00000000", "0?Set(REC_STATUS=NO)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:14] GotoIf("SIP/473-00000000", "5?checkaction") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (sub-record-check,s,17)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:17] GotoIf("SIP/473-00000000", "1?sub-record-check,exten,1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (sub-record-check,exten,1)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] NoOp("SIP/473-00000000", "Exten Recording Check between 473 and 446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Set("SIP/473-00000000", "CALLTYPE=internal") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] ExecIf("SIP/473-00000000", "0?Set(CALLTYPE=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] Set("SIP/473-00000000", "CALLEE=force") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:5] ExecIf("SIP/473-00000000", "0?Set(CALLEE=dontcare)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:6] GotoIf("SIP/473-00000000", "0?callee") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:7] GotoIf("SIP/473-00000000", "0?caller") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:8] ExecIf("SIP/473-00000000", "2?Set(CALLER_PRI=10):Set(CALLER_PRI=0)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:9] ExecIf("SIP/473-00000000", "2?Set(CALLEE_PRI=10):Set(CALLEE_PRI=0)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:10] GotoIf("SIP/473-00000000", "1?caller:callee") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (sub-record-check,exten,13)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:13] Set("SIP/473-00000000", "RECMODE=force") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:14] ExecIf("SIP/473-00000000", "0?Set(RECMODE=dontcare)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:15] ExecIf("SIP/473-00000000", "0?Set(RECMODE=force)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:16] Gosub("SIP/473-00000000", "recordcheck,1(force,internal,446)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] NoOp("SIP/473-00000000", "Starting recording check against force") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Goto("SIP/473-00000000", "force") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (sub-record-check,recordcheck,5)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:5] Set("SIP/473-00000000", "__REC_POLICY_MODE=FORCE") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:6] GotoIf("SIP/473-00000000", "1?startrec") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (sub-record-check,recordcheck,16)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:16] NoOp("SIP/473-00000000", "Starting recording: internal, 446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:17] Set("SIP/473-00000000", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
[2018-07-19 11:23:56] NOTICE[28305][C-00000000] func_audiohookinherit.c: AUDIOHOOK_INHERIT is deprecated and now does nothing.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:18] Set("SIP/473-00000000", "__CALLFILENAME=internal-446-473-20180719-112356-1531992236.0") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:19] MixMonitor("SIP/473-00000000", "2018/07/19/internal-446-473-20180719-112356-1531992236.0.wav,abi(LOCAL_MIXMON_ID),") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:20] Set("SIP/473-00000000", "__MIXMON_ID=0x9b04be8") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:21] Set("SIP/473-00000000", "__RECORD_ID=SIP/473-00000000") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:22] Set("SIP/473-00000000", "__REC_STATUS=RECORDING") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:23] Set("SIP/473-00000000", "CDR(recordingfile)=internal-446-473-20180719-112356-1531992236.0.wav") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:24] Return("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:17] Return("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:19] GotoIf("SIP/473-00000000", "1?macrodial") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-exten-vm,s,25)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:25] GosubIf("SIP/473-00000000", "0?clrheader,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:26] Macro("SIP/473-00000000", "dial-one,,Ttr,446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Set("SIP/473-00000000", "DEXTEN=446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] ExecIf("SIP/473-00000000", "0?Set(__EXTTOCALL=446)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] Set("SIP/473-00000000", "DIALSTATUS_CW=") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] GosubIf("SIP/473-00000000", "0?screen,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:5] GosubIf("SIP/473-00000000", "0?cf,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:6] GotoIf("SIP/473-00000000", "1?skip1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-dial-one,s,9)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:9] GotoIf("SIP/473-00000000", "0?nodial") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:10] GotoIf("SIP/473-00000000", "0?continue") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:11] Set("SIP/473-00000000", "EXTHASCW=ENABLED") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:12] GotoIf("SIP/473-00000000", "0?next1:cwinusebusy") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-dial-one,s,24)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:24] GotoIf("SIP/473-00000000", "0?next3:continue") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-dial-one,s,26)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:26] GotoIf("SIP/473-00000000", "0?nodial") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:27] GosubIf("SIP/473-00000000", "1?dstring,1():dlocal,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Set("SIP/473-00000000", "DSTRING=") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Set("SIP/473-00000000", "DEVICES=446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] ExecIf("SIP/473-00000000", "0?Return()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] ExecIf("SIP/473-00000000", "0?Set(DEVICES=46)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:5] Set("SIP/473-00000000", "LOOPCNT=1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:6] Set("SIP/473-00000000", "ITER=1") in new stack
[2018-07-19 11:23:56] VERBOSE[28311][C-00000000] app_mixmonitor.c: Begin MixMonitor Recording SIP/473-00000000
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:7] Set("SIP/473-00000000", "THISDIAL=SIP/446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:8] GosubIf("SIP/473-00000000", "1?zap2dahdi,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] ExecIf("SIP/473-00000000", "0?Return()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Set("SIP/473-00000000", "NEWDIAL=") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] Set("SIP/473-00000000", "LOOPCNT2=1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] Set("SIP/473-00000000", "ITER2=1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:5] Set("SIP/473-00000000", "THISPART2=SIP/446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:6] ExecIf("SIP/473-00000000", "0?Set(THISPART2=DAHDI/446)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:7] Set("SIP/473-00000000", "NEWDIAL=SIP/446&") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:8] Set("SIP/473-00000000", "ITER2=2") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:9] GotoIf("SIP/473-00000000", "0?begin2") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:10] Set("SIP/473-00000000", "THISDIAL=SIP/446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:11] Return("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:9] GotoIf("SIP/473-00000000", "1?docheck") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-dial-one,dstring,14)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:14] GotoIf("SIP/473-00000000", "0?skipset") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:15] Set("SIP/473-00000000", "DSTRING=SIP/446&") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:16] Set("SIP/473-00000000", "ITER=2") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:17] GotoIf("SIP/473-00000000", "0?begin") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:18] ExecIf("SIP/473-00000000", "0?Return()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:19] Set("SIP/473-00000000", "DSTRING=SIP/446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:20] Return("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:28] GotoIf("SIP/473-00000000", "0?nodial") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:29] GotoIf("SIP/473-00000000", "0?skiptrace") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:30] GosubIf("SIP/473-00000000", "1?ctset,1():ctclear,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Set("SIP/473-00000000", "DB(CALLTRACE/446)=473") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Return("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:31] Set("SIP/473-00000000", "D_OPTIONS=Ttr") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:32] GosubIf("SIP/473-00000000", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:33] NoOp("SIP/473-00000000", "Blind Transfer: , Attended Transfer: , User: 473, Alert Info: ") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:34] ExecIf("SIP/473-00000000", "1?Set(ALERT_INFO=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:35] ExecIf("SIP/473-00000000", "0?Set(ALERT_INFO=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:36] ExecIf("SIP/473-00000000", "0?Set(ALERT_INFO=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:37] ExecIf("SIP/473-00000000", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:38] ExecIf("SIP/473-00000000", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:39] GosubIf("SIP/473-00000000", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:40] ExecIf("SIP/473-00000000", "0?Set(CHANNEL(musicclass)=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:41] GosubIf("SIP/473-00000000", "0?qwait,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:42] Set("SIP/473-00000000", "__CWIGNORE=") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:43] Set("SIP/473-00000000", "__KEEPCID=TRUE") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:44] GotoIf("SIP/473-00000000", "0?usegoto,1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:45] GotoIf("SIP/473-00000000", "0?godial") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:46] Gosub("SIP/473-00000000", "sub-presencestate-display,s,1(446)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Goto("SIP/473-00000000", "state-not_set,1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (sub-presencestate-display,state-not_set,1)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Set("SIP/473-00000000", "PRESENCESTATE_DISPLAY=") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Return("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:47] Set("SIP/473-00000000", "CONNECTEDLINE(name,i)=446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:48] Set("SIP/473-00000000", "CONNECTEDLINE(num)=446") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:49] Set("SIP/473-00000000", "D_OPTIONS=TtrI") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:50] Macro("SIP/473-00000000", "dialout-one-predial-hook,") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] MacroExit("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:51] ExecIf("SIP/473-00000000", "0?Set(D_OPTIONS=trII)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:52] NoOp("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:53] ExecIf("SIP/473-00000000", "0?Set(D_OPTIONS=TtrIg)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:54] Dial("SIP/473-00000000", "SIP/446,,TtrIb(func-apply-sipheaders^s^1)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] netsock2.c: Using SIP RTP TOS bits 184
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] netsock2.c: Using SIP RTP CoS mark 5
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_stack.c: SIP/446-00000001 Internal Gosub(func-apply-sipheaders,s,1) start
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] NoOp("SIP/446-00000001", "Applying SIP Headers to channel") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] Set("SIP/446-00000001", "SIPHEADERKEYS=") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] ExecIf("SIP/446-00000001", "0?Set(Rheader=1)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] While("SIP/446-00000001", "0") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_while.c: Jumping to priority 8
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:9] ExecIf("SIP/446-00000001", "0?SIPRemoveHeader(Alert-Info:)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:10] ExecIf("SIP/446-00000001", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:11] Return("SIP/446-00000001", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_stack.c: Spawn extension (from-internal, 446, 1) exited non-zero on 'SIP/446-00000001'
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_stack.c: SIP/446-00000001 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_dial.c: Called SIP/446
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as140f1f52
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_dial.c: Connected line update to SIP/473-00000000 prevented.
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:55] ExecIf("SIP/473-00000000", "0?MacroExit()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:56] ExecIf("SIP/473-00000000", "0?Set(DIALSTATUS=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:57] GosubIf("SIP/473-00000000", "0?s-CHANUNAVAIL,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:58] MacroExit("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:27] Set("SIP/473-00000000", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:28] GosubIf("SIP/473-00000000", "0?docfu,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:29] GosubIf("SIP/473-00000000", "0?docfb,1()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:30] Set("SIP/473-00000000", "DIALSTATUS=CHANUNAVAIL") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:31] ExecIf("SIP/473-00000000", "0?MacroExit()") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:32] GotoIf("SIP/473-00000000", "1?s-CHANUNAVAIL,1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-exten-vm,s-CHANUNAVAIL,1)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] GotoIf("SIP/473-00000000", "0?exit,1") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:2] PlayTones("SIP/473-00000000", "congestion") in new stack
[2018-07-19 11:23:56] WARNING[28305][C-00000000] translate.c: no samples for ulawtolin
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] Congestion("SIP/473-00000000", "10") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as140f1f52
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


<------------>
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_macro.c: Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/473-00000000' in macro 'exten-vm'
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Spawn extension (ext-local, 446, 2) exited non-zero on 'SIP/473-00000000'
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] Macro("SIP/473-00000000", "hangupcall,") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:1] GotoIf("SIP/473-00000000", "1?theend") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:3] ExecIf("SIP/473-00000000", "0?Set(CDR(recordingfile)=)") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Executing [[email protected]:4] Hangup("SIP/473-00000000", "") in new stack
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/473-00000000' in macro 'hangupcall'
[2018-07-19 11:23:56] VERBOSE[28305][C-00000000] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/473-00000000'
[2018-07-19 11:23:56] VERBOSE[28311][C-00000000] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2018-07-19 11:23:56] VERBOSE[28311][C-00000000] app_mixmonitor.c: End MixMonitor Recording SIP/473-00000000
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK11076960
From: "473" <sip:[email protected]:5060>;tag=2677010622
To: <sip:[email protected]:5060>;tag=as140f1f52
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

<------------->
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c: --- (7 headers 0 lines) ---
[2018-07-19 11:23:56] VERBOSE[21329] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: ACK
[2018-07-19 11:24:12] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 11:24:12] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 11:24:12] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 11:24:20] NOTICE[21329] chan_sip.c:    -- Re-registration for  xxx
[2018-07-19 11:24:20] NOTICE[21329] chan_sip.c: Outbound Registration: Expiry for xxx is 96 sec (Scheduling reregistration in 81 s)
[2018-07-19 11:24:33] NOTICE[21329] chan_sip.c:    -- Re-registration for  xxx
[2018-07-19 11:24:33] NOTICE[21329] chan_sip.c: Outbound Registration: Expiry for xxx is 111 sec (Scheduling reregistration in 96 s)
[2018-07-19 11:24:38] VERBOSE[21329] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.129:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3af0e0bb
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as1f7d4b5c
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.195.4(13.19.1)
Date: Thu, 19 Jul 2018 09:24:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2018-07-19 11:24:38] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3af0e0bb
From: "Unknown" <sip:[email protected]>;tag=as1f7d4b5c
To: <sip:[email protected]:5060>;tag=2257710922
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Content-Length: 0

<------------->
[2018-07-19 11:24:38] VERBOSE[21329] chan_sip.c: --- (8 headers 0 lines) ---
[2018-07-19 11:24:38] VERBOSE[21329] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2018-07-19 11:24:42] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 11:24:42] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 11:24:42] VERBOSE[21329] chan_sip.c:
<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 11:24:48] VERBOSE[26596] asterisk.c: Remote UNIX connection disconnected




[Ovu poruku je menjao bogdan.kecman dana 19.07.2018. u 12:39 GMT+1]
[ bogdan.kecman @ 19.07.2018. 10:35 ] @
sad ne vidim ono 404 sto sam video juce, cak vidim da pise ringing ali telefon nije zvonio
[ bogdan.kecman @ 19.07.2018. 11:32 ] @
jos jedan pokusaj

[code]

117 sip peers [Monitored: 95 online, 21 offline Unmonitored: 1 online, 0 offline]

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 12:28:15] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:28:15] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:28:15] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:28:15] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:28:19] NOTICE[21329]: chan_sip.c:15716 sip_reregister: -- Re-registration for xxx
[2018-07-19 12:28:19] NOTICE[21329]: chan_sip.c:24551 handle_response_register: Outbound Registration: Expiry for xxx is 104 sec (Scheduling reregistration in 89 s)
Reliably Transmitting (no NAT) to 192.168.1.129:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5c954555
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as214b356f
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.195.4(13.19.1)
Date: Thu, 19 Jul 2018 10:28:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5c954555
From: "Unknown" <sip:[email protected]>;tag=as214b356f
To: <sip:[email protected]:5060>;tag=1198134991
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2018-07-19 12:28:27] NOTICE[21329]: chan_sip.c:15716 sip_reregister: -- Re-registration for xxx
[2018-07-19 12:28:27] NOTICE[21329]: chan_sip.c:24551 handle_response_register: Outbound Registration: Expiry for xxx is 92 sec (Scheduling reregistration in 77 s)

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
Reliably Transmitting (no NAT) to 192.168.1.129:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0f93e244
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as52d81a9d
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.195.4(13.19.1)
Date: Thu, 19 Jul 2018 10:29:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0f93e244
From: "Unknown" <sip:[email protected]>;tag=as52d81a9d
To: <sip:[email protected]:5060>;tag=2822357359
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK4059750581
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21761 21761 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12594 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.1.129:5060 (NAT)
Sending to 192.168.1.129:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK4059750581;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as5028fc87
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49f61a14"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK4059750581
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as5028fc87
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="473", realm="asterisk", nonce="49f61a14", uri="sip:[email protected]:5060", response="565f055ff31f5fbe901c3377752b697a", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21761 21761 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12594 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.1.129:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0xb7624160 -- Strict RTP learning after remote address set to: 192.168.1.129:12594
Peer audio RTP is at port 192.168.1.129:12594
Looking for 446 in from-internal (domain 192.168.1.10)
sip_route_dump: route/path hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
-- Executing [[email protected]:1] GotoIf("SIP/473-00000002", "1?ext-local,446,1:followme-check,446,1") in new stack
-- Goto (ext-local,446,1)
-- Executing [[email protected]:1] Set("SIP/473-00000002", "__RINGTIMER=15") in new stack
-- Executing [[email protected]:2] Macro("SIP/473-00000002", "exten-vm,novm,446,0,0,0") in new stack
-- Executing [[email protected]:1] Macro("SIP/473-00000002", "user-callerid,") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000002", "TOUCH_MONITOR=1531996165.2") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "AMPUSER=473") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/473-00000002", "0?report") in new stack
-- Executing [[email protected]d:4] ExecIf("SIP/473-00000002", "1?Set(REALCALLERIDNUM=473)") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "AMPUSER=473") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000002", "0?limit") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000002", "AMPUSERCIDNAME=473") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000002", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000002", "0?report") in new stack
-- Executing [[email protected]:10] Set("SIP/473-00000002", "AMPUSERCID=473") in new stack
-- Executing [[email protected]:11] Set("SIP/473-00000002", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:12] Set("SIP/473-00000002", "CALLERID(all)="473" <473>") in new stack
-- Executing [[email protected]:13] GotoIf("SIP/473-00000002", "0?limit") in new stack
-- Executing [[email protected]:14] ExecIf("SIP/473-00000002", "0?Set(GROUP(concurrency_limit)=473)") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/473-00000002", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [[email protected]:16] NoOp("SIP/473-00000002", "Macro Depth is 2") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/473-00000002", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [[email protected]:18] GotoIf("SIP/473-00000002", "0?continue") in new stack
-- Executing [[email protected]:19] ExecIf("SIP/473-00000002", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
-- Executing [[email protected]:20] Set("SIP/473-00000002", "__TTL=64") in new stack
-- Executing [[email protected]:21] GotoIf("SIP/473-00000002", "1?continue") in new stack
-- Goto (macro-user-callerid,s,37)
-- Executing [[email protected]:37] Set("SIP/473-00000002", "CALLERID(number)=473") in new stack
-- Executing [[email protected]:38] Set("SIP/473-00000002", "CALLERID(name)=473") in new stack
-- Executing [[email protected]:39] GotoIf("SIP/473-00000002", "0?cnum") in new stack
-- Executing [[email protected]:40] Set("SIP/473-00000002", "CDR(cnam)=473") in new stack
-- Executing [[email protected]:41] Set("SIP/473-00000002", "CDR(cnum)=473") in new stack
-- Executing [[email protected]:42] Set("SIP/473-00000002", "CHANNEL(language)=en") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "RingGroupMethod=none") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000002", "__EXTTOCALL=446") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000002", "__PICKUPMARK=446") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "RT=") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/473-00000002", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
-- Executing [[email protected]:7] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000002", "0?Gosub(ext-intercom,*80446,1())") in new stack
-- Executing [[email protected]:9] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/473-00000002", "0?ChanSpy(SIP/446,q)") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:12] ExecIf("SIP/473-00000002", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:13] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:14] ExecIf("SIP/473-00000002", "0?Gosub(ext-intercom,*80446,1())") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:15] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:16] ExecIf("SIP/473-00000002", "0?ChanSpy(SIP/446,q)") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:17] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:18] Gosub("SIP/473-00000002", "sub-record-check,s,1(exten,446,dontcare)") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/473-00000002", "0?initialized") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000002", "NOW=1531996165") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000002", "__DAY=19") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "__MONTH=07") in new stack
-- Executing [[email protected]:6] Set("SIP/473-00000002", "__YEAR=2018") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000002", "__TIMESTR=20180719-122925") in new stack
-- Executing [[email protected]:8] Set("SIP/473-00000002", "__FROMEXTEN=473") in new stack
-- Executing [[email protected]:9] Set("SIP/473-00000002", "__MON_FMT=wav") in new stack
-- Executing [[email protected]:10] NoOp("SIP/473-00000002", "Recordings initialized") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/473-00000002", "0?Set(ARG3=dontcare)") in new stack
-- Executing [[email protected]:12] Set("SIP/473-00000002", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [[email protected]:13] ExecIf("SIP/473-00000002", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [[email protected]:14] GotoIf("SIP/473-00000002", "5?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [[email protected]:17] GotoIf("SIP/473-00000002", "1?sub-record-check,exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [[email protected]:1] NoOp("SIP/473-00000002", "Exten Recording Check between 473 and 446") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "CALLTYPE=internal") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/473-00000002", "0?Set(CALLTYPE=)") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000002", "CALLEE=force") in new stack
-- Executing [[email protected]:5] ExecIf("SIP/473-00000002", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000002", "0?callee") in new stack
-- Executing [[email protected]:7] GotoIf("SIP/473-00000002", "0?caller") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000002", "2?Set(CALLER_PRI=10):Set(CALLER_PRI=0)") in new stack
-- Executing [[email protected]:9] ExecIf("SIP/473-00000002", "2?Set(CALLEE_PRI=10):Set(CALLEE_PRI=0)") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/473-00000002", "1?caller:callee") in new stack
-- Goto (sub-record-check,exten,13)
-- Executing [[email protected]:13] Set("SIP/473-00000002", "RECMODE=force") in new stack
-- Executing [[email protected]:14] ExecIf("SIP/473-00000002", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/473-00000002", "0?Set(RECMODE=force)") in new stack
-- Executing [[email protected]:16] Gosub("SIP/473-00000002", "recordcheck,1(force,internal,446)") in new stack
-- Executing [[email protected]:1] NoOp("SIP/473-00000002", "Starting recording check against force") in new stack
-- Executing [[email protected]:2] Goto("SIP/473-00000002", "force") in new stack
-- Goto (sub-record-check,recordcheck,5)
-- Executing [[email protected]:5] Set("SIP/473-00000002", "__REC_POLICY_MODE=FORCE") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000002", "1?startrec") in new stack
-- Goto (sub-record-check,recordcheck,16)
-- Executing [[email protected]:16] NoOp("SIP/473-00000002", "Starting recording: internal, 446") in new stack
-- Executing [[email protected]:17] Set("SIP/473-00000002", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [[email protected]:18] Set("SIP/473-00000002", "__CALLFILENAME=internal-446-473-20180719-122925-1531996165.2") in new stack
-- Executing [[email protected]:19] MixMonitor("SIP/473-00000002", "2018/07/19/internal-446-473-20180719-122925-1531996165.2.wav,abi(LOCAL_MIXMON_ID),") in new stack
-- Executing [[email protected]:20] Set("SIP/473-00000002", "__MIXMON_ID=0x9ef7290") in new stack
-- Executing [[email protected]:21] Set("SIP/473-00000002", "__RECORD_ID=SIP/473-00000002") in new stack
-- Executing [[email protected]:22] Set("SIP/473-00000002", "__REC_STATUS=RECORDING") in new stack
-- Executing [[email protected]:23] Set("SIP/473-00000002", "CDR(recordingfile)=internal-446-473-20180719-122925-1531996165.2.wav") in new stack
-- Executing [[email protected]:24] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:17] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:19] GotoIf("SIP/473-00000002", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,25)
-- Executing [[email protected]:25] GosubIf("SIP/473-00000002", "0?clrheader,1()") in new stack
-- Executing [[email protected]:26] Macro("SIP/473-00000002", "dial-one,,Ttr,446") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000002", "DEXTEN=446") in new stack
-- Executing [[email protected]:2] ExecIf("SIP/473-00000002", "0?Set(__EXTTOCALL=446)") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000002", "DIALSTATUS_CW=") in new stack
-- Executing [[email protected]:4] GosubIf("SIP/473-00000002", "0?screen,1()") in new stack
-- Executing [[email protected]:5] GosubIf("SIP/473-00000002", "0?cf,1()") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000002", "1?skip1") in new stack
-- Goto (macro-dial-one,s,9)
-- Executing [[email protected]:9] GotoIf("SIP/473-00000002", "0?nodial") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/473-00000002", "0?continue") in new stack
-- Executing [[email protected]:11] Set("SIP/473-00000002", "EXTHASCW=ENABLED") in new stack
-- Executing [[email protected]:12] GotoIf("SIP/473-00000002", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,24)
-- Executing [[email protected]:24] GotoIf("SIP/473-00000002", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,26)
-- Executing [[email protected]:26] GotoIf("SIP/473-00000002", "0?nodial") in new stack
-- Executing [[email protected]:27] GosubIf("SIP/473-00000002", "1?dstring,1():dlocal,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000002", "DSTRING=") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "DEVICES=446") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/473-00000002", "0?Return()") in new stack
-- Executing [[email protected]:4] ExecIf("SIP/473-00000002", "0?Set(DEVICES=46)") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "LOOPCNT=1") in new stack
-- Executing [[email protected]:6] Set("SIP/473-00000002", "ITER=1") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000002", "THISDIAL=SIP/446") in new stack
-- Executing [[email protected]:8] GosubIf("SIP/473-00000002", "1?zap2dahdi,1()") in new stack
-- Executing [[email protected]:1] ExecIf("SIP/473-00000002", "0?Return()") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "NEWDIAL=") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000002", "LOOPCNT2=1") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000002", "ITER2=1") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "THISPART2=SIP/446") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/473-00000002", "0?Set(THISPART2=DAHDI/446)") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000002", "NEWDIAL=SIP/446&") in new stack
-- Executing [[email protected]:8] Set("SIP/473-00000002", "ITER2=2") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000002", "0?begin2") in new stack
-- Executing [[email protected]:10] Set("SIP/473-00000002", "THISDIAL=SIP/446") in new stack
-- Executing [[email protected]:11] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000002", "1?docheck") in new stack
-- Goto (macro-dial-one,dstring,14)
-- Executing [[email protected]:14] GotoIf("SIP/473-00000002", "0?skipset") in new stack
-- Executing [[email protected]:15] Set("SIP/473-00000002", "DSTRING=SIP/446&") in new stack
-- Executing [[email protected]:16] Set("SIP/473-00000002", "ITER=2") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/473-00000002", "0?begin") in new stack
-- Executing [[email protected]:18] ExecIf("SIP/473-00000002", "0?Return()") in new stack
== Begin MixMonitor Recording SIP/473-00000002
-- Executing [[email protected]:19] Set("SIP/473-00000002", "DSTRING=SIP/446") in new stack
-- Executing [[email protected]:20] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:28] GotoIf("SIP/473-00000002", "0?nodial") in new stack
-- Executing [[email protected]:29] GotoIf("SIP/473-00000002", "0?skiptrace") in new stack
-- Executing [[email protected]:30] GosubIf("SIP/473-00000002", "1?ctset,1():ctclear,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000002", "DB(CALLTRACE/446)=473") in new stack
-- Executing [[email protected]:2] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:31] Set("SIP/473-00000002", "D_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:32] GosubIf("SIP/473-00000002", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [[email protected]:33] NoOp("SIP/473-00000002", "Blind Transfer: , Attended Transfer: , User: 473, Alert Info: ") in new stack
-- Executing [[email protected]:34] ExecIf("SIP/473-00000002", "1?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:35] ExecIf("SIP/473-00000002", "0?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:36] ExecIf("SIP/473-00000002", "0?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:37] ExecIf("SIP/473-00000002", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [[email protected]:38] ExecIf("SIP/473-00000002", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [[email protected]:39] GosubIf("SIP/473-00000002", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [[email protected]:40] ExecIf("SIP/473-00000002", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [[email protected]:41] GosubIf("SIP/473-00000002", "0?qwait,1()") in new stack
-- Executing [[email protected]:42] Set("SIP/473-00000002", "__CWIGNORE=") in new stack
-- Executing [[email protected]:43] Set("SIP/473-00000002", "__KEEPCID=TRUE") in new stack
-- Executing [[email protected]:44] GotoIf("SIP/473-00000002", "0?usegoto,1") in new stack
-- Executing [[email protected]:45] GotoIf("SIP/473-00000002", "0?godial") in new stack
-- Executing [[email protected]:46] Gosub("SIP/473-00000002", "sub-presencestate-display,s,1(446)") in new stack
-- Executing [[email protected]:1] Goto("SIP/473-00000002", "state-not_set,1") in new stack
-- Goto (sub-presencestate-display,state-not_set,1)
-- Executing [[email protected]:1] Set("SIP/473-00000002", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [[email protected]:2] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:47] Set("SIP/473-00000002", "CONNECTEDLINE(name,i)=446") in new stack
-- Executing [[email protected]:48] Set("SIP/473-00000002", "CONNECTEDLINE(num)=446") in new stack
-- Executing [[email protected]:49] Set("SIP/473-00000002", "D_OPTIONS=TtrI") in new stack
-- Executing [[email protected]:50] Macro("SIP/473-00000002", "dialout-one-predial-hook,") in new stack
-- Executing [[email protected]:1] MacroExit("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:51] ExecIf("SIP/473-00000002", "0?Set(D_OPTIONS=trII)") in new stack
-- Executing [[email protected]:52] NoOp("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:53] ExecIf("SIP/473-00000002", "0?Set(D_OPTIONS=TtrIg)") in new stack
-- Executing [[email protected]:54] Dial("SIP/473-00000002", "SIP/446,,TtrIb(func-apply-sipheaders^s^1)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- SIP/446-00000003 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [[email protected]:1] NoOp("SIP/446-00000003", "Applying SIP Headers to channel") in new stack
-- Executing [[email protected]:2] Set("SIP/446-00000003", "SIPHEADERKEYS=") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/446-00000003", "0?Set(Rheader=1)") in new stack
-- Executing [[email protected]:4] While("SIP/446-00000003", "0") in new stack
-- Jumping to priority 8
-- Executing [[email protected]:9] ExecIf("SIP/446-00000003", "0?SIPRemoveHeader(Alert-Info:)") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/446-00000003", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
-- Executing [[email protected]:11] Return("SIP/446-00000003", "") in new stack
== Spawn extension (from-internal, 446, 1) exited non-zero on 'SIP/446-00000003'
-- SIP/446-00000003 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called SIP/446

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as397088eb
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
-- Connected line update to SIP/473-00000002 prevented.
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [[email protected]:55] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
-- Executing [[email protected]:56] ExecIf("SIP/473-00000002", "0?Set(DIALSTATUS=)") in new stack
-- Executing [[email protected]:57] GosubIf("SIP/473-00000002", "0?s-CHANUNAVAIL,1()") in new stack
-- Executing [[email protected]:58] MacroExit("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:27] Set("SIP/473-00000002", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:28] GosubIf("SIP/473-00000002", "0?docfu,1()") in new stack
-- Executing [[email protected]:29] GosubIf("SIP/473-00000002", "0?docfb,1()") in new stack
-- Executing [[email protected]:30] Set("SIP/473-00000002", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:31] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
-- Executing [[email protected]:32] GotoIf("SIP/473-00000002", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [[email protected]:1] GotoIf("SIP/473-00000002", "0?exit,1") in new stack
-- Executing [[email protected]:2] PlayTones("SIP/473-00000002", "congestion") in new stack
[2018-07-19 12:29:25] WARNING[26592][C-00000001]: translate.c:407 framein: no samples for ulawtolin
-- Executing [[email protected]:3] Congestion("SIP/473-00000002", "10") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as397088eb
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/473-00000002' in macro 'exten-vm'
== Spawn extension (ext-local, 446, 2) exited non-zero on 'SIP/473-00000002'
-- Executing [[email protected]:1] Macro("SIP/473-00000002", "hangupcall,") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/473-00000002", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [[email protected]:3] ExecIf("SIP/473-00000002", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [[email protected]:4] Hangup("SIP/473-00000002", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/473-00000002' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/473-00000002'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/473-00000002

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as397088eb
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK1545217739
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21762 21762 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12596 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.1.129:5060 (NAT)
Sending to 192.168.1.129:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK1545217739;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as48477fb4
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36656da5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK1545217739
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as48477fb4
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="473", realm="asterisk", nonce="36656da5", uri="sip:[email protected]:5060", response="44214a65425157e1d058890ff6dd1f95", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21762 21762 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12596 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.1.129:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0xb7624160 -- Strict RTP learning after remote address set to: 192.168.1.129:12596
Peer audio RTP is at port 192.168.1.129:12596
Looking for 446 in from-internal (domain 192.168.1.10)
sip_route_dump: route/path hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
-- Executing [[email protected]:1] GotoIf("SIP/473-00000004", "1?ext-local,446,1:followme-check,446,1") in new stack
-- Goto (ext-local,446,1)
-- Executing [[email protected]:1] Set("SIP/473-00000004", "__RINGTIMER=15") in new stack
-- Executing [[email protected]:2] Macro("SIP/473-00000004", "exten-vm,novm,446,0,0,0") in new stack
-- Executing [[email protected]:1] Macro("SIP/473-00000004", "user-callerid,") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000004", "TOUCH_MONITOR=1531996174.4") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "AMPUSER=473") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/473-00000004", "0?report") in new stack
-- Executing [[email protected]:4] ExecIf("SIP/473-00000004", "1?Set(REALCALLERIDNUM=473)") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "AMPUSER=473") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000004", "0?limit") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000004", "AMPUSERCIDNAME=473") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000004", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000004", "0?report") in new stack
-- Executing [[email protected]:10] Set("SIP/473-00000004", "AMPUSERCID=473") in new stack
-- Executing [[email protected]:11] Set("SIP/473-00000004", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:12] Set("SIP/473-00000004", "CALLERID(all)="473" <473>") in new stack
-- Executing [[email protected]:13] GotoIf("SIP/473-00000004", "0?limit") in new stack
-- Executing [[email protected]:14] ExecIf("SIP/473-00000004", "0?Set(GROUP(concurrency_limit)=473)") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/473-00000004", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [[email protected]:16] NoOp("SIP/473-00000004", "Macro Depth is 2") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/473-00000004", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [[email protected]:18] GotoIf("SIP/473-00000004", "0?continue") in new stack
-- Executing [[email protected]:19] ExecIf("SIP/473-00000004", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
-- Executing [[email protected]:20] Set("SIP/473-00000004", "__TTL=64") in new stack
-- Executing [[email protected]:21] GotoIf("SIP/473-00000004", "1?continue") in new stack
-- Goto (macro-user-callerid,s,37)
-- Executing [[email protected]:37] Set("SIP/473-00000004", "CALLERID(number)=473") in new stack
-- Executing [[email protected]:38] Set("SIP/473-00000004", "CALLERID(name)=473") in new stack
-- Executing [[email protected]:39] GotoIf("SIP/473-00000004", "0?cnum") in new stack
-- Executing [[email protected]:40] Set("SIP/473-00000004", "CDR(cnam)=473") in new stack
-- Executing [[email protected]:41] Set("SIP/473-00000004", "CDR(cnum)=473") in new stack
-- Executing [[email protected]:42] Set("SIP/473-00000004", "CHANNEL(language)=en") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "RingGroupMethod=none") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000004", "__EXTTOCALL=446") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000004", "__PICKUPMARK=446") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "RT=") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/473-00000004", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
-- Executing [[email protected]:7] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000004", "0?Gosub(ext-intercom,*80446,1())") in new stack
-- Executing [[email protected]:9] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/473-00000004", "0?ChanSpy(SIP/446,q)") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:12] ExecIf("SIP/473-00000004", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:13] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:14] ExecIf("SIP/473-00000004", "0?Gosub(ext-intercom,*80446,1())") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:15] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:16] ExecIf("SIP/473-00000004", "0?ChanSpy(SIP/446,q)") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:17] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:18] Gosub("SIP/473-00000004", "sub-record-check,s,1(exten,446,dontcare)") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/473-00000004", "0?initialized") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000004", "NOW=1531996174") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000004", "__DAY=19") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "__MONTH=07") in new stack
-- Executing [[email protected]:6] Set("SIP/473-00000004", "__YEAR=2018") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000004", "__TIMESTR=20180719-122934") in new stack
-- Executing [[email protected]:8] Set("SIP/473-00000004", "__FROMEXTEN=473") in new stack
-- Executing [[email protected]:9] Set("SIP/473-00000004", "__MON_FMT=wav") in new stack
-- Executing [[email protected]:10] NoOp("SIP/473-00000004", "Recordings initialized") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/473-00000004", "0?Set(ARG3=dontcare)") in new stack
-- Executing [[email protected]:12] Set("SIP/473-00000004", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [[email protected]:13] ExecIf("SIP/473-00000004", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [[email protected]:14] GotoIf("SIP/473-00000004", "5?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [[email protected]:17] GotoIf("SIP/473-00000004", "1?sub-record-check,exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [[email protected]:1] NoOp("SIP/473-00000004", "Exten Recording Check between 473 and 446") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "CALLTYPE=internal") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/473-00000004", "0?Set(CALLTYPE=)") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000004", "CALLEE=force") in new stack
-- Executing [[email protected]:5] ExecIf("SIP/473-00000004", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000004", "0?callee") in new stack
-- Executing [[email protected]:7] GotoIf("SIP/473-00000004", "0?caller") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000004", "2?Set(CALLER_PRI=10):Set(CALLER_PRI=0)") in new stack
-- Executing [[email protected]:9] ExecIf("SIP/473-00000004", "2?Set(CALLEE_PRI=10):Set(CALLEE_PRI=0)") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/473-00000004", "1?caller:callee") in new stack
-- Goto (sub-record-check,exten,13)
-- Executing [[email protected]:13] Set("SIP/473-00000004", "RECMODE=force") in new stack
-- Executing [[email protected]:14] ExecIf("SIP/473-00000004", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/473-00000004", "0?Set(RECMODE=force)") in new stack
-- Executing [[email protected]:16] Gosub("SIP/473-00000004", "recordcheck,1(force,internal,446)") in new stack
-- Executing [[email protected]:1] NoOp("SIP/473-00000004", "Starting recording check against force") in new stack
-- Executing [[email protected]:2] Goto("SIP/473-00000004", "force") in new stack
-- Goto (sub-record-check,recordcheck,5)
-- Executing [[email protected]:5] Set("SIP/473-00000004", "__REC_POLICY_MODE=FORCE") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000004", "1?startrec") in new stack
-- Goto (sub-record-check,recordcheck,16)
-- Executing [[email protected]:16] NoOp("SIP/473-00000004", "Starting recording: internal, 446") in new stack
-- Executing [[email protected]:17] Set("SIP/473-00000004", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [[email protected]:18] Set("SIP/473-00000004", "__CALLFILENAME=internal-446-473-20180719-122934-1531996174.4") in new stack
-- Executing [[email protected]:19] MixMonitor("SIP/473-00000004", "2018/07/19/internal-446-473-20180719-122934-1531996174.4.wav,abi(LOCAL_MIXMON_ID),") in new stack
-- Executing [[email protected]:20] Set("SIP/473-00000004", "__MIXMON_ID=0x9ef7290") in new stack
-- Executing [[email protected]:21] Set("SIP/473-00000004", "__RECORD_ID=SIP/473-00000004") in new stack
-- Executing [[email protected]:22] Set("SIP/473-00000004", "__REC_STATUS=RECORDING") in new stack
-- Executing [[email protected]:23] Set("SIP/473-00000004", "CDR(recordingfile)=internal-446-473-20180719-122934-1531996174.4.wav") in new stack
-- Executing [[email protected]:24] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:17] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:19] GotoIf("SIP/473-00000004", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,25)
-- Executing [[email protected]:25] GosubIf("SIP/473-00000004", "0?clrheader,1()") in new stack
-- Executing [[email protected]:26] Macro("SIP/473-00000004", "dial-one,,Ttr,446") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000004", "DEXTEN=446") in new stack
-- Executing [[email protected]:2] ExecIf("SIP/473-00000004", "0?Set(__EXTTOCALL=446)") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000004", "DIALSTATUS_CW=") in new stack
-- Executing [[email protected]:4] GosubIf("SIP/473-00000004", "0?screen,1()") in new stack
-- Executing [[email protected]:5] GosubIf("SIP/473-00000004", "0?cf,1()") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000004", "1?skip1") in new stack
-- Goto (macro-dial-one,s,9)
-- Executing [[email protected]:9] GotoIf("SIP/473-00000004", "0?nodial") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/473-00000004", "0?continue") in new stack
-- Executing [[email protected]:11] Set("SIP/473-00000004", "EXTHASCW=ENABLED") in new stack
-- Executing [[email protected]:12] GotoIf("SIP/473-00000004", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,24)
-- Executing [[email protected]:24] GotoIf("SIP/473-00000004", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,26)
-- Executing [[email protected]:26] GotoIf("SIP/473-00000004", "0?nodial") in new stack
-- Executing [[email protected]:27] GosubIf("SIP/473-00000004", "1?dstring,1():dlocal,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000004", "DSTRING=") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "DEVICES=446") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/473-00000004", "0?Return()") in new stack
-- Executing [[email protected]:4] ExecIf("SIP/473-00000004", "0?Set(DEVICES=46)") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "LOOPCNT=1") in new stack
-- Executing [[email protected]:6] Set("SIP/473-00000004", "ITER=1") in new stack
== Begin MixMonitor Recording SIP/473-00000004
-- Executing [[email protected]:7] Set("SIP/473-00000004", "THISDIAL=SIP/446") in new stack
-- Executing [[email protected]:8] GosubIf("SIP/473-00000004", "1?zap2dahdi,1()") in new stack
-- Executing [[email protected]:1] ExecIf("SIP/473-00000004", "0?Return()") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "NEWDIAL=") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000004", "LOOPCNT2=1") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000004", "ITER2=1") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "THISPART2=SIP/446") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/473-00000004", "0?Set(THISPART2=DAHDI/446)") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000004", "NEWDIAL=SIP/446&") in new stack
-- Executing [[email protected]:8] Set("SIP/473-00000004", "ITER2=2") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000004", "0?begin2") in new stack
-- Executing [[email protected]:10] Set("SIP/473-00000004", "THISDIAL=SIP/446") in new stack
-- Executing [[email protected]:11] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000004", "1?docheck") in new stack
-- Goto (macro-dial-one,dstring,14)
-- Executing [[email protected]:14] GotoIf("SIP/473-00000004", "0?skipset") in new stack
-- Executing [[email protected]:15] Set("SIP/473-00000004", "DSTRING=SIP/446&") in new stack
-- Executing [[email protected]:16] Set("SIP/473-00000004", "ITER=2") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/473-00000004", "0?begin") in new stack
-- Executing [[email protected]:18] ExecIf("SIP/473-00000004", "0?Return()") in new stack
-- Executing [[email protected]:19] Set("SIP/473-00000004", "DSTRING=SIP/446") in new stack
-- Executing [[email protected]:20] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:28] GotoIf("SIP/473-00000004", "0?nodial") in new stack
-- Executing [[email protected]:29] GotoIf("SIP/473-00000004", "0?skiptrace") in new stack
-- Executing [[email protected]:30] GosubIf("SIP/473-00000004", "1?ctset,1():ctclear,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000004", "DB(CALLTRACE/446)=473") in new stack
-- Executing [[email protected]:2] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:31] Set("SIP/473-00000004", "D_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:32] GosubIf("SIP/473-00000004", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [[email protected]:33] NoOp("SIP/473-00000004", "Blind Transfer: , Attended Transfer: , User: 473, Alert Info: ") in new stack
-- Executing [[email protected]:34] ExecIf("SIP/473-00000004", "1?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:35] ExecIf("SIP/473-00000004", "0?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:36] ExecIf("SIP/473-00000004", "0?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:37] ExecIf("SIP/473-00000004", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [[email protected]:38] ExecIf("SIP/473-00000004", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [[email protected]:39] GosubIf("SIP/473-00000004", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [[email protected]:40] ExecIf("SIP/473-00000004", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [[email protected]:41] GosubIf("SIP/473-00000004", "0?qwait,1()") in new stack
-- Executing [[email protected]:42] Set("SIP/473-00000004", "__CWIGNORE=") in new stack
-- Executing [[email protected]:43] Set("SIP/473-00000004", "__KEEPCID=TRUE") in new stack
-- Executing [[email protected]:44] GotoIf("SIP/473-00000004", "0?usegoto,1") in new stack
-- Executing [[email protected]:45] GotoIf("SIP/473-00000004", "0?godial") in new stack
-- Executing [[email protected]:46] Gosub("SIP/473-00000004", "sub-presencestate-display,s,1(446)") in new stack
-- Executing [[email protected]:1] Goto("SIP/473-00000004", "state-not_set,1") in new stack
-- Goto (sub-presencestate-display,state-not_set,1)
-- Executing [[email protected]:1] Set("SIP/473-00000004", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [[email protected]:2] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:47] Set("SIP/473-00000004", "CONNECTEDLINE(name,i)=446") in new stack
-- Executing [[email protected]:48] Set("SIP/473-00000004", "CONNECTEDLINE(num)=446") in new stack
-- Executing [[email protected]:49] Set("SIP/473-00000004", "D_OPTIONS=TtrI") in new stack
-- Executing [[email protected]:50] Macro("SIP/473-00000004", "dialout-one-predial-hook,") in new stack
-- Executing [[email protected]:1] MacroExit("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:51] ExecIf("SIP/473-00000004", "0?Set(D_OPTIONS=trII)") in new stack
-- Executing [[email protected]:52] NoOp("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:53] ExecIf("SIP/473-00000004", "0?Set(D_OPTIONS=TtrIg)") in new stack
-- Executing [[email protected]:54] Dial("SIP/473-00000004", "SIP/446,,TtrIb(func-apply-sipheaders^s^1)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- SIP/446-00000005 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [[email protected]:1] NoOp("SIP/446-00000005", "Applying SIP Headers to channel") in new stack
-- Executing [[email protected]:2] Set("SIP/446-00000005", "SIPHEADERKEYS=") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/446-00000005", "0?Set(Rheader=1)") in new stack
-- Executing [[email protected]:4] While("SIP/446-00000005", "0") in new stack
-- Jumping to priority 8
-- Executing [[email protected]:9] ExecIf("SIP/446-00000005", "0?SIPRemoveHeader(Alert-Info:)") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/446-00000005", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
-- Executing [[email protected]:11] Return("SIP/446-00000005", "") in new stack
== Spawn extension (from-internal, 446, 1) exited non-zero on 'SIP/446-00000005'
-- SIP/446-00000005 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called SIP/446

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as6783110c
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
-- Connected line update to SIP/473-00000004 prevented.
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [[email protected]:55] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
-- Executing [[email protected]:56] ExecIf("SIP/473-00000004", "0?Set(DIALSTATUS=)") in new stack
-- Executing [[email protected]:57] GosubIf("SIP/473-00000004", "0?s-CHANUNAVAIL,1()") in new stack
-- Executing [[email protected]:58] MacroExit("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:27] Set("SIP/473-00000004", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:28] GosubIf("SIP/473-00000004", "0?docfu,1()") in new stack
-- Executing [[email protected]:29] GosubIf("SIP/473-00000004", "0?docfb,1()") in new stack
-- Executing [[email protected]:30] Set("SIP/473-00000004", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:31] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
-- Executing [[email protected]:32] GotoIf("SIP/473-00000004", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [[email protected]:1] GotoIf("SIP/473-00000004", "0?exit,1") in new stack
-- Executing [[email protected]:2] PlayTones("SIP/473-00000004", "congestion") in new stack
[2018-07-19 12:29:34] WARNING[27023][C-00000002]: translate.c:407 framein: no samples for ulawtolin
-- Executing [[email protected]:3] Congestion("SIP/473-00000004", "10") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as6783110c
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/473-00000004' in macro 'exten-vm'
== Spawn extension (ext-local, 446, 2) exited non-zero on 'SIP/473-00000004'
-- Executing [[email protected]:1] Macro("SIP/473-00000004", "hangupcall,") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/473-00000004", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as6783110c
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
-- Executing [[email protected]:3] ExecIf("SIP/473-00000004", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [[email protected]:4] Hangup("SIP/473-00000004", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/473-00000004' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/473-00000004'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/473-00000004
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 12:29:44] NOTICE[21329]: chan_sip.c:15716 sip_reregister: -- Re-registration for xxx
[2018-07-19 12:29:45] NOTICE[21329]: chan_sip.c:24551 handle_response_register: Outbound Registration: Expiry for xxx is 86 sec (Scheduling reregistration in 71 s)
[2018-07-19 12:29:48] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:29:48] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:29:48] NOTICE[21329]: chan_sip.c:15716 sip_reregister: -- Re-registration for xxxm
[2018-07-19 12:29:49] NOTICE[21329]: chan_sip.c:24551 handle_response_register: Outbound Registration: Expiry for xxx is 101 sec (Scheduling reregistration in 86 s)
[ bogdan.kecman @ 19.07.2018. 11:36 ] @
p[ojede ovaj gore log

[code]

117 sip peers [Monitored: 95 online, 21 offline Unmonitored: 1 online, 0 offline]

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
[2018-07-19 12:28:15] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:28:15] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:28:15] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:28:15] NOTICE[21329]: chan_sip.c:30207 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2018-07-19 12:28:19] NOTICE[21329]: chan_sip.c:15716 sip_reregister: -- Re-registration for xxx
[2018-07-19 12:28:19] NOTICE[21329]: chan_sip.c:24551 handle_response_register: Outbound Registration: Expiry for xxx is 104 sec (Scheduling reregistration in 89 s)
Reliably Transmitting (no NAT) to 192.168.1.129:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5c954555
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as214b356f
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.195.4(13.19.1)
Date: Thu, 19 Jul 2018 10:28:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5c954555
From: "Unknown" <sip:[email protected]>;tag=as214b356f
To: <sip:[email protected]:5060>;tag=1198134991
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2018-07-19 12:28:27] NOTICE[21329]: chan_sip.c:15716 sip_reregister: -- Re-registration for xxx
[2018-07-19 12:28:27] NOTICE[21329]: chan_sip.c:24551 handle_response_register: Outbound Registration: Expiry for xxx is 92 sec (Scheduling reregistration in 77 s)

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


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

<--- SIP read from UDP:192.168.1.129:5060 --->


<------------->
Reliably Transmitting (no NAT) to 192.168.1.129:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0f93e244
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as52d81a9d
To: <sip:[email protected]:5060>
Contact: <sip:U[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.195.4(13.19.1)
Date: Thu, 19 Jul 2018 10:29:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0f93e244
From: "Unknown" <sip:[email protected]>;tag=as52d81a9d
To: <sip:[email protected]:5060>;tag=2822357359
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK4059750581
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21761 21761 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12594 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.1.129:5060 (NAT)
Sending to 192.168.1.129:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK4059750581;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as5028fc87
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49f61a14"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK4059750581
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as5028fc87
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="473", realm="asterisk", nonce="49f61a14", uri="sip:[email protected]:5060", response="565f055ff31f5fbe901c3377752b697a", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21761 21761 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12594 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.1.129:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0xb7624160 -- Strict RTP learning after remote address set to: 192.168.1.129:12594
Peer audio RTP is at port 192.168.1.129:12594
Looking for 446 in from-internal (domain 192.168.1.10)
sip_route_dump: route/path hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
-- Executing [[email protected]:1] GotoIf("SIP/473-00000002", "1?ext-local,446,1:followme-check,446,1") in new stack
-- Goto (ext-local,446,1)
-- Executing [[email protected]:1] Set("SIP/473-00000002", "__RINGTIMER=15") in new stack
-- Executing [[email protected]:2] Macro("SIP/473-00000002", "exten-vm,novm,446,0,0,0") in new stack
-- Executing [[email protected]:1] Macro("SIP/473-00000002", "user-callerid,") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000002", "TOUCH_MONITOR=1531996165.2") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "AMPUSER=473") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/473-00000002", "0?report") in new stack
-- Executing [[email protected]:4] ExecIf("SIP/473-00000002", "1?Set(REALCALLERIDNUM=473)") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "AMPUSER=473") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000002", "0?limit") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000002", "AMPUSERCIDNAME=473") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000002", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000002", "0?report") in new stack
-- Executing [[email protected]:10] Set("SIP/473-00000002", "AMPUSERCID=473") in new stack
-- Executing [[email protected]:11] Set("SIP/473-00000002", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:12] Set("SIP/473-00000002", "CALLERID(all)="473" <473>") in new stack
-- Executing [[email protected]:13] GotoIf("SIP/473-00000002", "0?limit") in new stack
-- Executing [[email protected]:14] ExecIf("SIP/473-00000002", "0?Set(GROUP(concurrency_limit)=473)") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/473-00000002", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [[email protected]:16] NoOp("SIP/473-00000002", "Macro Depth is 2") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/473-00000002", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [[email protected]:18] GotoIf("SIP/473-00000002", "0?continue") in new stack
-- Executing [[email protected]:19] ExecIf("SIP/473-00000002", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
-- Executing [[email protected]:20] Set("SIP/473-00000002", "__TTL=64") in new stack
-- Executing [[email protected]:21] GotoIf("SIP/473-00000002", "1?continue") in new stack
-- Goto (macro-user-callerid,s,37)
-- Executing [[email protected]:37] Set("SIP/473-00000002", "CALLERID(number)=473") in new stack
-- Executing [[email protected]:38] Set("SIP/473-00000002", "CALLERID(name)=473") in new stack
-- Executing [[email protected]:39] GotoIf("SIP/473-00000002", "0?cnum") in new stack
-- Executing [[email protected]:40] Set("SIP/473-00000002", "CDR(cnam)=473") in new stack
-- Executing [[email protected]:41] Set("SIP/473-00000002", "CDR(cnum)=473") in new stack
-- Executing [[email protected]:42] Set("SIP/473-00000002", "CHANNEL(language)=en") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "RingGroupMethod=none") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000002", "__EXTTOCALL=446") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000002", "__PICKUPMARK=446") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "RT=") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/473-00000002", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
-- Executing [[email protected]:7] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000002", "0?Gosub(ext-intercom,*80446,1())") in new stack
-- Executing [[email protected]exten-vm:9] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/473-00000002", "0?ChanSpy(SIP/446,q)") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:12] ExecIf("SIP/473-00000002", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:13] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:14] ExecIf("SIP/473-00000002", "0?Gosub(ext-intercom,*80446,1())") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:15] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:16] ExecIf("SIP/473-00000002", "0?ChanSpy(SIP/446,q)") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:17] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
[2018-07-19 12:29:25] ERROR[26592][C-00000001]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:18] Gosub("SIP/473-00000002", "sub-record-check,s,1(exten,446,dontcare)") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/473-00000002", "0?initialized") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000002", "NOW=1531996165") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000002", "__DAY=19") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "__MONTH=07") in new stack
-- Executing [[email protected]:6] Set("SIP/473-00000002", "__YEAR=2018") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000002", "__TIMESTR=20180719-122925") in new stack
-- Executing [[email protected]:8] Set("SIP/473-00000002", "__FROMEXTEN=473") in new stack
-- Executing [[email protected]:9] Set("SIP/473-00000002", "__MON_FMT=wav") in new stack
-- Executing [[email protected]:10] NoOp("SIP/473-00000002", "Recordings initialized") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/473-00000002", "0?Set(ARG3=dontcare)") in new stack
-- Executing [[email protected]:12] Set("SIP/473-00000002", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [[email protected]:13] ExecIf("SIP/473-00000002", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [[email protected]:14] GotoIf("SIP/473-00000002", "5?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [[email protected]:17] GotoIf("SIP/473-00000002", "1?sub-record-check,exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [[email protected]:1] NoOp("SIP/473-00000002", "Exten Recording Check between 473 and 446") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "CALLTYPE=internal") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/473-00000002", "0?Set(CALLTYPE=)") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000002", "CALLEE=force") in new stack
-- Executing [[email protected]:5] ExecIf("SIP/473-00000002", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000002", "0?callee") in new stack
-- Executing [[email protected]:7] GotoIf("SIP/473-00000002", "0?caller") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000002", "2?Set(CALLER_PRI=10):Set(CALLER_PRI=0)") in new stack
-- Executing [[email protected]:9] ExecIf("SIP/473-00000002", "2?Set(CALLEE_PRI=10):Set(CALLEE_PRI=0)") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/473-00000002", "1?caller:callee") in new stack
-- Goto (sub-record-check,exten,13)
-- Executing [[email protected]:13] Set("SIP/473-00000002", "RECMODE=force") in new stack
-- Executing [[email protected]:14] ExecIf("SIP/473-00000002", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/473-00000002", "0?Set(RECMODE=force)") in new stack
-- Executing [[email protected]:16] Gosub("SIP/473-00000002", "recordcheck,1(force,internal,446)") in new stack
-- Executing [[email protected]:1] NoOp("SIP/473-00000002", "Starting recording check against force") in new stack
-- Executing [[email protected]:2] Goto("SIP/473-00000002", "force") in new stack
-- Goto (sub-record-check,recordcheck,5)
-- Executing [[email protected]:5] Set("SIP/473-00000002", "__REC_POLICY_MODE=FORCE") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000002", "1?startrec") in new stack
-- Goto (sub-record-check,recordcheck,16)
-- Executing [[email protected]:16] NoOp("SIP/473-00000002", "Starting recording: internal, 446") in new stack
-- Executing [[email protected]:17] Set("SIP/473-00000002", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [[email protected]:18] Set("SIP/473-00000002", "__CALLFILENAME=internal-446-473-20180719-122925-1531996165.2") in new stack
-- Executing [[email protected]:19] MixMonitor("SIP/473-00000002", "2018/07/19/internal-446-473-20180719-122925-1531996165.2.wav,abi(LOCAL_MIXMON_ID),") in new stack
-- Executing [[email protected]:20] Set("SIP/473-00000002", "__MIXMON_ID=0x9ef7290") in new stack
-- Executing [[email protected]:21] Set("SIP/473-00000002", "__RECORD_ID=SIP/473-00000002") in new stack
-- Executing [[email protected]:22] Set("SIP/473-00000002", "__REC_STATUS=RECORDING") in new stack
-- Executing [[email protected]:23] Set("SIP/473-00000002", "CDR(recordingfile)=internal-446-473-20180719-122925-1531996165.2.wav") in new stack
-- Executing [[email protected]:24] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:17] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:19] GotoIf("SIP/473-00000002", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,25)
-- Executing [[email protected]:25] GosubIf("SIP/473-00000002", "0?clrheader,1()") in new stack
-- Executing [[email protected]:26] Macro("SIP/473-00000002", "dial-one,,Ttr,446") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000002", "DEXTEN=446") in new stack
-- Executing [[email protected]:2] ExecIf("SIP/473-00000002", "0?Set(__EXTTOCALL=446)") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000002", "DIALSTATUS_CW=") in new stack
-- Executing [[email protected]:4] GosubIf("SIP/473-00000002", "0?screen,1()") in new stack
-- Executing [[email protected]:5] GosubIf("SIP/473-00000002", "0?cf,1()") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000002", "1?skip1") in new stack
-- Goto (macro-dial-one,s,9)
-- Executing [[email protected]:9] GotoIf("SIP/473-00000002", "0?nodial") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/473-00000002", "0?continue") in new stack
-- Executing [[email protected]:11] Set("SIP/473-00000002", "EXTHASCW=ENABLED") in new stack
-- Executing [[email protected]:12] GotoIf("SIP/473-00000002", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,24)
-- Executing [[email protected]:24] GotoIf("SIP/473-00000002", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,26)
-- Executing [[email protected]:26] GotoIf("SIP/473-00000002", "0?nodial") in new stack
-- Executing [[email protected]:27] GosubIf("SIP/473-00000002", "1?dstring,1():dlocal,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000002", "DSTRING=") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "DEVICES=446") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/473-00000002", "0?Return()") in new stack
-- Executing [[email protected]:4] ExecIf("SIP/473-00000002", "0?Set(DEVICES=46)") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "LOOPCNT=1") in new stack
-- Executing [[email protected]:6] Set("SIP/473-00000002", "ITER=1") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000002", "THISDIAL=SIP/446") in new stack
-- Executing [[email protected]:8] GosubIf("SIP/473-00000002", "1?zap2dahdi,1()") in new stack
-- Executing [[email protected]one:1] ExecIf("SIP/473-00000002", "0?Return()") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000002", "NEWDIAL=") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000002", "LOOPCNT2=1") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000002", "ITER2=1") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000002", "THISPART2=SIP/446") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/473-00000002", "0?Set(THISPART2=DAHDI/446)") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000002", "NEWDIAL=SIP/446&") in new stack
-- Executing [[email protected]:8] Set("SIP/473-00000002", "ITER2=2") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000002", "0?begin2") in new stack
-- Executing [[email protected]:10] Set("SIP/473-00000002", "THISDIAL=SIP/446") in new stack
-- Executing [[email protected]:11] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000002", "1?docheck") in new stack
-- Goto (macro-dial-one,dstring,14)
-- Executing [[email protected]:14] GotoIf("SIP/473-00000002", "0?skipset") in new stack
-- Executing [[email protected]:15] Set("SIP/473-00000002", "DSTRING=SIP/446&") in new stack
-- Executing [[email protected]:16] Set("SIP/473-00000002", "ITER=2") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/473-00000002", "0?begin") in new stack
-- Executing [[email protected]:18] ExecIf("SIP/473-00000002", "0?Return()") in new stack
== Begin MixMonitor Recording SIP/473-00000002
-- Executing [[email protected]:19] Set("SIP/473-00000002", "DSTRING=SIP/446") in new stack
-- Executing [[email protected]:20] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:28] GotoIf("SIP/473-00000002", "0?nodial") in new stack
-- Executing [[email protected]:29] GotoIf("SIP/473-00000002", "0?skiptrace") in new stack
-- Executing [[email protected]:30] GosubIf("SIP/473-00000002", "1?ctset,1():ctclear,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000002", "DB(CALLTRACE/446)=473") in new stack
-- Executing [[email protected]:2] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:31] Set("SIP/473-00000002", "D_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:32] GosubIf("SIP/473-00000002", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [[email protected]:33] NoOp("SIP/473-00000002", "Blind Transfer: , Attended Transfer: , User: 473, Alert Info: ") in new stack
-- Executing [[email protected]:34] ExecIf("SIP/473-00000002", "1?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:35] ExecIf("SIP/473-00000002", "0?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:36] ExecIf("SIP/473-00000002", "0?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:37] ExecIf("SIP/473-00000002", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [[email protected]:38] ExecIf("SIP/473-00000002", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [[email protected]:39] GosubIf("SIP/473-00000002", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [[email protected]:40] ExecIf("SIP/473-00000002", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [[email protected]:41] GosubIf("SIP/473-00000002", "0?qwait,1()") in new stack
-- Executing [[email protected]:42] Set("SIP/473-00000002", "__CWIGNORE=") in new stack
-- Executing [[email protected]:43] Set("SIP/473-00000002", "__KEEPCID=TRUE") in new stack
-- Executing [[email protected]:44] GotoIf("SIP/473-00000002", "0?usegoto,1") in new stack
-- Executing [[email protected]:45] GotoIf("SIP/473-00000002", "0?godial") in new stack
-- Executing [[email protected]:46] Gosub("SIP/473-00000002", "sub-presencestate-display,s,1(446)") in new stack
-- Executing [[email protected]:1] Goto("SIP/473-00000002", "state-not_set,1") in new stack
-- Goto (sub-presencestate-display,state-not_set,1)
-- Executing [[email protected]:1] Set("SIP/473-00000002", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [[email protected]:2] Return("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:47] Set("SIP/473-00000002", "CONNECTEDLINE(name,i)=446") in new stack
-- Executing [[email protected]:48] Set("SIP/473-00000002", "CONNECTEDLINE(num)=446") in new stack
-- Executing [[email protected]:49] Set("SIP/473-00000002", "D_OPTIONS=TtrI") in new stack
-- Executing [[email protected]:50] Macro("SIP/473-00000002", "dialout-one-predial-hook,") in new stack
-- Executing [[email protected]:1] MacroExit("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:51] ExecIf("SIP/473-00000002", "0?Set(D_OPTIONS=trII)") in new stack
-- Executing [[email protected]:52] NoOp("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:53] ExecIf("SIP/473-00000002", "0?Set(D_OPTIONS=TtrIg)") in new stack
-- Executing [[email protected]:54] Dial("SIP/473-00000002", "SIP/446,,TtrIb(func-apply-sipheaders^s^1)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- SIP/446-00000003 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [[email protected]:1] NoOp("SIP/446-00000003", "Applying SIP Headers to channel") in new stack
-- Executing [[email protected]:2] Set("SIP/446-00000003", "SIPHEADERKEYS=") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/446-00000003", "0?Set(Rheader=1)") in new stack
-- Executing [[email protected]:4] While("SIP/446-00000003", "0") in new stack
-- Jumping to priority 8
-- Executing [[email protected]:9] ExecIf("SIP/446-00000003", "0?SIPRemoveHeader(Alert-Info:)") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/446-00000003", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
-- Executing [[email protected]:11] Return("SIP/446-00000003", "") in new stack
== Spawn extension (from-internal, 446, 1) exited non-zero on 'SIP/446-00000003'
-- SIP/446-00000003 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called SIP/446

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as397088eb
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
-- Connected line update to SIP/473-00000002 prevented.
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [[email protected]:55] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
-- Executing [[email protected]:56] ExecIf("SIP/473-00000002", "0?Set(DIALSTATUS=)") in new stack
-- Executing [[email protected]:57] GosubIf("SIP/473-00000002", "0?s-CHANUNAVAIL,1()") in new stack
-- Executing [[email protected]:58] MacroExit("SIP/473-00000002", "") in new stack
-- Executing [[email protected]:27] Set("SIP/473-00000002", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:28] GosubIf("SIP/473-00000002", "0?docfu,1()") in new stack
-- Executing [[email protected]:29] GosubIf("SIP/473-00000002", "0?docfb,1()") in new stack
-- Executing [[email protected]:30] Set("SIP/473-00000002", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:31] ExecIf("SIP/473-00000002", "0?MacroExit()") in new stack
-- Executing [[email protected]:32] GotoIf("SIP/473-00000002", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [[email protected]:1] GotoIf("SIP/473-00000002", "0?exit,1") in new stack
-- Executing [[email protected]:2] PlayTones("SIP/473-00000002", "congestion") in new stack
[2018-07-19 12:29:25] WARNING[26592][C-00000001]: translate.c:407 framein: no samples for ulawtolin
-- Executing [[email protected]:3] Congestion("SIP/473-00000002", "10") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as397088eb
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/473-00000002' in macro 'exten-vm'
== Spawn extension (ext-local, 446, 2) exited non-zero on 'SIP/473-00000002'
-- Executing [[email protected]:1] Macro("SIP/473-00000002", "hangupcall,") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/473-00000002", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [[email protected]:3] ExecIf("SIP/473-00000002", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [[email protected]:4] Hangup("SIP/473-00000002", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/473-00000002' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/473-00000002'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/473-00000002

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3612345516
From: "473" <sip:[email protected]:5060>;tag=3423495360
To: <sip:[email protected]:5060>;tag=as397088eb
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK1545217739
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21762 21762 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12596 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.1.129:5060 (NAT)
Sending to 192.168.1.129:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK1545217739;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as48477fb4
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36656da5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK1545217739
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as48477fb4
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.129:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="473", realm="asterisk", nonce="36656da5", uri="sip:[email protected]:5060", response="44214a65425157e1d058890ff6dd1f95", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.95
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21762 21762 IN IP4 192.168.1.129
s=SDP data
c=IN IP4 192.168.1.129
t=0 0
m=audio 12596 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.1.129:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '473' for '473' from 192.168.1.129:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0xb7624160 -- Strict RTP learning after remote address set to: 192.168.1.129:12596
Peer audio RTP is at port 192.168.1.129:12596
Looking for 446 in from-internal (domain 192.168.1.10)
sip_route_dump: route/path hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
-- Executing [[email protected]:1] GotoIf("SIP/473-00000004", "1?ext-local,446,1:followme-check,446,1") in new stack
-- Goto (ext-local,446,1)
-- Executing [[email protected]:1] Set("SIP/473-00000004", "__RINGTIMER=15") in new stack
-- Executing [[email protected]:2] Macro("SIP/473-00000004", "exten-vm,novm,446,0,0,0") in new stack
-- Executing [[email protected]:1] Macro("SIP/473-00000004", "user-callerid,") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000004", "TOUCH_MONITOR=1531996174.4") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "AMPUSER=473") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/473-00000004", "0?report") in new stack
-- Executing [[email protected]:4] ExecIf("SIP/473-00000004", "1?Set(REALCALLERIDNUM=473)") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "AMPUSER=473") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000004", "0?limit") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000004", "AMPUSERCIDNAME=473") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000004", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000004", "0?report") in new stack
-- Executing [[email protected]:10] Set("SIP/473-00000004", "AMPUSERCID=473") in new stack
-- Executing [[email protected]:11] Set("SIP/473-00000004", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:12] Set("SIP/473-00000004", "CALLERID(all)="473" <473>") in new stack
-- Executing [[email protected]:13] GotoIf("SIP/473-00000004", "0?limit") in new stack
-- Executing [[email protected]:14] ExecIf("SIP/473-00000004", "0?Set(GROUP(concurrency_limit)=473)") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/473-00000004", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [[email protected]:16] NoOp("SIP/473-00000004", "Macro Depth is 2") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/473-00000004", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [[email protected]:18] GotoIf("SIP/473-00000004", "0?continue") in new stack
-- Executing [[email protected]:19] ExecIf("SIP/473-00000004", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
-- Executing [[email protected]:20] Set("SIP/473-00000004", "__TTL=64") in new stack
-- Executing [[email protected]:21] GotoIf("SIP/473-00000004", "1?continue") in new stack
-- Goto (macro-user-callerid,s,37)
-- Executing [[email protected]:37] Set("SIP/473-00000004", "CALLERID(number)=473") in new stack
-- Executing [[email protected]:38] Set("SIP/473-00000004", "CALLERID(name)=473") in new stack
-- Executing [[email protected]:39] GotoIf("SIP/473-00000004", "0?cnum") in new stack
-- Executing [[email protected]:40] Set("SIP/473-00000004", "CDR(cnam)=473") in new stack
-- Executing [[email protected]:41] Set("SIP/473-00000004", "CDR(cnum)=473") in new stack
-- Executing [[email protected]:42] Set("SIP/473-00000004", "CHANNEL(language)=en") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "RingGroupMethod=none") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000004", "__EXTTOCALL=446") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000004", "__PICKUPMARK=446") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "RT=") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/473-00000004", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
-- Executing [[email protected]:7] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000004", "0?Gosub(ext-intercom,*80446,1())") in new stack
-- Executing [[email protected]:9] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/473-00000004", "0?ChanSpy(SIP/446,q)") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:12] ExecIf("SIP/473-00000004", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:13] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:14] ExecIf("SIP/473-00000004", "0?Gosub(ext-intercom,*80446,1())") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:15] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:16] ExecIf("SIP/473-00000004", "0?ChanSpy(SIP/446,q)") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:17] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
[2018-07-19 12:29:34] ERROR[27023][C-00000002]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [[email protected]:18] Gosub("SIP/473-00000004", "sub-record-check,s,1(exten,446,dontcare)") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/473-00000004", "0?initialized") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000004", "NOW=1531996174") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000004", "__DAY=19") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "__MONTH=07") in new stack
-- Executing [[email protected]:6] Set("SIP/473-00000004", "__YEAR=2018") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000004", "__TIMESTR=20180719-122934") in new stack
-- Executing [[email protected]:8] Set("SIP/473-00000004", "__FROMEXTEN=473") in new stack
-- Executing [[email protected]:9] Set("SIP/473-00000004", "__MON_FMT=wav") in new stack
-- Executing [[email protected]:10] NoOp("SIP/473-00000004", "Recordings initialized") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/473-00000004", "0?Set(ARG3=dontcare)") in new stack
-- Executing [[email protected]:12] Set("SIP/473-00000004", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [[email protected]:13] ExecIf("SIP/473-00000004", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [[email protected]:14] GotoIf("SIP/473-00000004", "5?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [[email protected]:17] GotoIf("SIP/473-00000004", "1?sub-record-check,exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [[email protected]:1] NoOp("SIP/473-00000004", "Exten Recording Check between 473 and 446") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "CALLTYPE=internal") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/473-00000004", "0?Set(CALLTYPE=)") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000004", "CALLEE=force") in new stack
-- Executing [[email protected]:5] ExecIf("SIP/473-00000004", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000004", "0?callee") in new stack
-- Executing [[email protected]:7] GotoIf("SIP/473-00000004", "0?caller") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/473-00000004", "2?Set(CALLER_PRI=10):Set(CALLER_PRI=0)") in new stack
-- Executing [[email protected]:9] ExecIf("SIP/473-00000004", "2?Set(CALLEE_PRI=10):Set(CALLEE_PRI=0)") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/473-00000004", "1?caller:callee") in new stack
-- Goto (sub-record-check,exten,13)
-- Executing [[email protected]:13] Set("SIP/473-00000004", "RECMODE=force") in new stack
-- Executing [[email protected]:14] ExecIf("SIP/473-00000004", "0?Set(RECMODE=dontcare)") in new stack
-- Executing [[email protected]:15] ExecIf("SIP/473-00000004", "0?Set(RECMODE=force)") in new stack
-- Executing [[email protected]:16] Gosub("SIP/473-00000004", "recordcheck,1(force,internal,446)") in new stack
-- Executing [[email protected]:1] NoOp("SIP/473-00000004", "Starting recording check against force") in new stack
-- Executing [[email protected]:2] Goto("SIP/473-00000004", "force") in new stack
-- Goto (sub-record-check,recordcheck,5)
-- Executing [[email protected]:5] Set("SIP/473-00000004", "__REC_POLICY_MODE=FORCE") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000004", "1?startrec") in new stack
-- Goto (sub-record-check,recordcheck,16)
-- Executing [[email protected]:16] NoOp("SIP/473-00000004", "Starting recording: internal, 446") in new stack
-- Executing [[email protected]:17] Set("SIP/473-00000004", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [[email protected]:18] Set("SIP/473-00000004", "__CALLFILENAME=internal-446-473-20180719-122934-1531996174.4") in new stack
-- Executing [[email protected]:19] MixMonitor("SIP/473-00000004", "2018/07/19/internal-446-473-20180719-122934-1531996174.4.wav,abi(LOCAL_MIXMON_ID),") in new stack
-- Executing [[email protected]:20] Set("SIP/473-00000004", "__MIXMON_ID=0x9ef7290") in new stack
-- Executing [[email protected]:21] Set("SIP/473-00000004", "__RECORD_ID=SIP/473-00000004") in new stack
-- Executing [[email protected]:22] Set("SIP/473-00000004", "__REC_STATUS=RECORDING") in new stack
-- Executing [[email protected]:23] Set("SIP/473-00000004", "CDR(recordingfile)=internal-446-473-20180719-122934-1531996174.4.wav") in new stack
-- Executing [[email protected]:24] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:17] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:19] GotoIf("SIP/473-00000004", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,25)
-- Executing [[email protected]:25] GosubIf("SIP/473-00000004", "0?clrheader,1()") in new stack
-- Executing [[email protected]:26] Macro("SIP/473-00000004", "dial-one,,Ttr,446") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000004", "DEXTEN=446") in new stack
-- Executing [[email protected]:2] ExecIf("SIP/473-00000004", "0?Set(__EXTTOCALL=446)") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000004", "DIALSTATUS_CW=") in new stack
-- Executing [[email protected]:4] GosubIf("SIP/473-00000004", "0?screen,1()") in new stack
-- Executing [[email protected]:5] GosubIf("SIP/473-00000004", "0?cf,1()") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/473-00000004", "1?skip1") in new stack
-- Goto (macro-dial-one,s,9)
-- Executing [[email protected]:9] GotoIf("SIP/473-00000004", "0?nodial") in new stack
-- Executing [[email protected]:10] GotoIf("SIP/473-00000004", "0?continue") in new stack
-- Executing [[email protected]:11] Set("SIP/473-00000004", "EXTHASCW=ENABLED") in new stack
-- Executing [[email protected]:12] GotoIf("SIP/473-00000004", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,24)
-- Executing [[email protected]:24] GotoIf("SIP/473-00000004", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,26)
-- Executing [[email protected]:26] GotoIf("SIP/473-00000004", "0?nodial") in new stack
-- Executing [[email protected]:27] GosubIf("SIP/473-00000004", "1?dstring,1():dlocal,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000004", "DSTRING=") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "DEVICES=446") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/473-00000004", "0?Return()") in new stack
-- Executing [[email protected]:4] ExecIf("SIP/473-00000004", "0?Set(DEVICES=46)") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "LOOPCNT=1") in new stack
-- Executing [[email protected]:6] Set("SIP/473-00000004", "ITER=1") in new stack
== Begin MixMonitor Recording SIP/473-00000004
-- Executing [[email protected]:7] Set("SIP/473-00000004", "THISDIAL=SIP/446") in new stack
-- Executing [[email protected]:8] GosubIf("SIP/473-00000004", "1?zap2dahdi,1()") in new stack
-- Executing [[email protected]:1] ExecIf("SIP/473-00000004", "0?Return()") in new stack
-- Executing [[email protected]:2] Set("SIP/473-00000004", "NEWDIAL=") in new stack
-- Executing [[email protected]:3] Set("SIP/473-00000004", "LOOPCNT2=1") in new stack
-- Executing [[email protected]:4] Set("SIP/473-00000004", "ITER2=1") in new stack
-- Executing [[email protected]:5] Set("SIP/473-00000004", "THISPART2=SIP/446") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/473-00000004", "0?Set(THISPART2=DAHDI/446)") in new stack
-- Executing [[email protected]:7] Set("SIP/473-00000004", "NEWDIAL=SIP/446&") in new stack
-- Executing [[email protected]:8] Set("SIP/473-00000004", "ITER2=2") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000004", "0?begin2") in new stack
-- Executing [[email protected]:10] Set("SIP/473-00000004", "THISDIAL=SIP/446") in new stack
-- Executing [[email protected]:11] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/473-00000004", "1?docheck") in new stack
-- Goto (macro-dial-one,dstring,14)
-- Executing [[email protected]:14] GotoIf("SIP/473-00000004", "0?skipset") in new stack
-- Executing [[email protected]:15] Set("SIP/473-00000004", "DSTRING=SIP/446&") in new stack
-- Executing [[email protected]:16] Set("SIP/473-00000004", "ITER=2") in new stack
-- Executing [[email protected]:17] GotoIf("SIP/473-00000004", "0?begin") in new stack
-- Executing [[email protected]:18] ExecIf("SIP/473-00000004", "0?Return()") in new stack
-- Executing [[email protected]:19] Set("SIP/473-00000004", "DSTRING=SIP/446") in new stack
-- Executing [[email protected]:20] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:28] GotoIf("SIP/473-00000004", "0?nodial") in new stack
-- Executing [[email protected]:29] GotoIf("SIP/473-00000004", "0?skiptrace") in new stack
-- Executing [[email protected]:30] GosubIf("SIP/473-00000004", "1?ctset,1():ctclear,1()") in new stack
-- Executing [[email protected]:1] Set("SIP/473-00000004", "DB(CALLTRACE/446)=473") in new stack
-- Executing [[email protected]:2] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:31] Set("SIP/473-00000004", "D_OPTIONS=Ttr") in new stack
-- Executing [[email protected]:32] GosubIf("SIP/473-00000004", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [[email protected]:33] NoOp("SIP/473-00000004", "Blind Transfer: , Attended Transfer: , User: 473, Alert Info: ") in new stack
-- Executing [[email protected]:34] ExecIf("SIP/473-00000004", "1?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:35] ExecIf("SIP/473-00000004", "0?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:36] ExecIf("SIP/473-00000004", "0?Set(ALERT_INFO=)") in new stack
-- Executing [[email protected]:37] ExecIf("SIP/473-00000004", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [[email protected]:38] ExecIf("SIP/473-00000004", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [[email protected]:39] GosubIf("SIP/473-00000004", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [[email protected]:40] ExecIf("SIP/473-00000004", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [[email protected]:41] GosubIf("SIP/473-00000004", "0?qwait,1()") in new stack
-- Executing [[email protected]:42] Set("SIP/473-00000004", "__CWIGNORE=") in new stack
-- Executing [[email protected]:43] Set("SIP/473-00000004", "__KEEPCID=TRUE") in new stack
-- Executing [[email protected]:44] GotoIf("SIP/473-00000004", "0?usegoto,1") in new stack
-- Executing [[email protected]:45] GotoIf("SIP/473-00000004", "0?godial") in new stack
-- Executing [[email protected]:46] Gosub("SIP/473-00000004", "sub-presencestate-display,s,1(446)") in new stack
-- Executing [[email protected]:1] Goto("SIP/473-00000004", "state-not_set,1") in new stack
-- Goto (sub-presencestate-display,state-not_set,1)
-- Executing [[email protected]:1] Set("SIP/473-00000004", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [[email protected]:2] Return("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:47] Set("SIP/473-00000004", "CONNECTEDLINE(name,i)=446") in new stack
-- Executing [[email protected]:48] Set("SIP/473-00000004", "CONNECTEDLINE(num)=446") in new stack
-- Executing [[email protected]:49] Set("SIP/473-00000004", "D_OPTIONS=TtrI") in new stack
-- Executing [[email protected]:50] Macro("SIP/473-00000004", "dialout-one-predial-hook,") in new stack
-- Executing [[email protected]:1] MacroExit("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:51] ExecIf("SIP/473-00000004", "0?Set(D_OPTIONS=trII)") in new stack
-- Executing [[email protected]:52] NoOp("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:53] ExecIf("SIP/473-00000004", "0?Set(D_OPTIONS=TtrIg)") in new stack
-- Executing [[email protected]:54] Dial("SIP/473-00000004", "SIP/446,,TtrIb(func-apply-sipheaders^s^1)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- SIP/446-00000005 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [[email protected]:1] NoOp("SIP/446-00000005", "Applying SIP Headers to channel") in new stack
-- Executing [[email protected]:2] Set("SIP/446-00000005", "SIPHEADERKEYS=") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/446-00000005", "0?Set(Rheader=1)") in new stack
-- Executing [[email protected]:4] While("SIP/446-00000005", "0") in new stack
-- Jumping to priority 8
-- Executing [[email protected]:9] ExecIf("SIP/446-00000005", "0?SIPRemoveHeader(Alert-Info:)") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/446-00000005", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
-- Executing [[email protected]:11] Return("SIP/446-00000005", "") in new stack
== Spawn extension (from-internal, 446, 1) exited non-zero on 'SIP/446-00000005'
-- SIP/446-00000005 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called SIP/446

<--- Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as6783110c
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
-- Connected line update to SIP/473-00000004 prevented.
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [[email protected]:55] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
-- Executing [[email protected]:56] ExecIf("SIP/473-00000004", "0?Set(DIALSTATUS=)") in new stack
-- Executing [[email protected]:57] GosubIf("SIP/473-00000004", "0?s-CHANUNAVAIL,1()") in new stack
-- Executing [[email protected]:58] MacroExit("SIP/473-00000004", "") in new stack
-- Executing [[email protected]:27] Set("SIP/473-00000004", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:28] GosubIf("SIP/473-00000004", "0?docfu,1()") in new stack
-- Executing [[email protected]:29] GosubIf("SIP/473-00000004", "0?docfb,1()") in new stack
-- Executing [[email protected]:30] Set("SIP/473-00000004", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [[email protected]:31] ExecIf("SIP/473-00000004", "0?MacroExit()") in new stack
-- Executing [[email protected]:32] GotoIf("SIP/473-00000004", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [[email protected]:1] GotoIf("SIP/473-00000004", "0?exit,1") in new stack
-- Executing [[email protected]:2] PlayTones("SIP/473-00000004", "congestion") in new stack
[2018-07-19 12:29:34] WARNING[27023][C-00000002]: translate.c:407 framein: no samples for ulawtolin
-- Executing [[email protected]:3] Congestion("SIP/473-00000004", "10") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.1.129:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507;received=192.168.1.129
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as6783110c
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-13.0.195.4(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/473-00000004' in macro 'exten-vm'
== Spawn extension (ext-local, 446, 2) exited non-zero on 'SIP/473-00000004'
-- Executing [[email protected]:1] Macro("SIP/473-00000004", "hangupcall,") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/473-00000004", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)

<--- SIP read from UDP:192.168.1.129:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK3486552507
From: "473" <sip:[email protected]:5060>;tag=594104001
To: <sip:[email protected]:5060>;tag=as6783110c
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
-- Executing [[email protected]:3] ExecIf("SIP/473-00000004", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [[email protected]:4] Hangup("SIP/473-00000004", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/473-00000004' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/473-00000004'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/473-00000004
Really destroying SIP dialog '[email protected]' Meth
[ bogdan.kecman @ 19.07.2018. 11:37 ] @
evo ovde ceo log

https://pastebin.com/s5tZszX3
[ valjan @ 19.07.2018. 12:52 ] @
Ako je taj lokal 446 postojao ranije, a s obzirom na to da freepbx dosta stvari generiše na osnovu podataka iz baze, a ne iz conf. fajlova kao kod klot asteriska, moguće da se u bazi pokarabasilo štogod, viđao sam slične probleme u matorim verzijama elastixa, dok je još koristio freepbx kao osnovu, a iskusio sam i slične probleme sa trankovima i rutama na par freepbx sistema koje sam održavao. U suštini obrišeš taj lokal, potamaniš sve relevantne zapise u bazi gde se pojavljuje lokal 446, pa ga kreiraš ponovo.
[ bogdan.kecman @ 19.07.2018. 13:12 ] @
pazi, taj 446 je bio, ali napravio sam 300 potpuno nov, nikad nije
postojao, 507 - 515 potpuno novi, nikad nisu postojali .. ista prica :(
[ valjan @ 19.07.2018. 14:19 ] @
Napisao si da se 446 registrovao sa IP adrese 192.168.1.204, međutim tog IP-ja nema nigde u onom logu koji si okačio na pastebin, a po pravilima SIP komunikacije morao bi se pojaviti u headeru u To i From polju (zavisno od toga ko šalje koji paket u kom trenutku). Inače i greška 404 i 503 uglavnom ukazuju na mrežni problem, u ovom slučaju najverovatnije nema SIP uređaja na adresi na kojoj je prijavljen (ili mu se ne može prići zbog nekog pravila koje blokira pristup)...
[ bogdan.kecman @ 19.07.2018. 14:27 ] @
lan je ok, ping radi, fw na  lanu je ugasen, arp kaze to je taj mac, sip
show peers kaze to je taj ip

kako nema u hederu:

od juce sa 404 je tu ... sad ga u novom do duse nema .... lan je sigurno
ok, a sad konfiguracija * ocigledno nije

Code:

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK783b3a7e
From: "475" ;tag=as5c07bd3b
To: ;tag=294278800
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T21P 34.72.0.75
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

[ bogdan.kecman @ 19.07.2018. 14:31 ] @
Code:

localhost*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
300                       (Unspecified)                            D  No         No          A  0        UNKNOWN
401/401                   192.168.1.118                            D  No         No          A  5062     OK (77 ms)
...
446/446                   192.168.1.204                            D  No         No          A  5065     OK (38 ms)
447/447                   192.168.1.186                            D  No         No          A  5062     OK (73 ms)
448                       (Unspecified)                            D  No         No          A  0        UNKNOWN
...
472                       (Unspecified)                            D  No         No          A  0        UNKNOWN
473/473                   192.168.1.129                            D  No         No          A  5060     OK (8 ms)
...
[ bogdan.kecman @ 19.07.2018. 14:34 ] @
ne konta, ove portove, to je verovatno port na telefonu? posto je centrala samo 5060 ?
[ bogdan.kecman @ 19.07.2018. 14:37 ] @
verovatno je udp port ali hm na 473 radi...

Code:

[[email protected] tftpboot]# telnet 192.168.1.204 5065
Trying 192.168.1.204...
telnet: connect to address 192.168.1.204: Connection refused
[[email protected] tftpboot]# nmap 192.168.1.204

Starting Nmap 5.51 ( http://nmap.org ) at 2018-07-19 15:35 CEST
Nmap scan report for 192.168.1.204
Host is up (0.0042s latency).
Not shown: 998 closed ports
PORT    STATE SERVICE
80/tcp  open  http
443/tcp open  https
MAC Address: 00:15:65:79:4D:54 (Xiamen Yealink Network Technology Co.)

Nmap done: 1 IP address (1 host up) scanned in 0.28 seconds
[[email protected] tftpboot]# telnet 192.168.1.129 5060
Trying 192.168.1.129...
Connected to 192.168.1.129.
Escape character is '^]'.
lkjl
^]
telnet> c
Connection closed.
[[email protected] tftpboot]#

[ bogdan.kecman @ 19.07.2018. 15:55 ] @
nista je123no nisam menjao i sad 446 zvoni :( :( ... pa zapalicu ga
[ bogdan.kecman @ 19.07.2018. 18:00 ] @
ostali naravno ne rade ..

evo ceo log veliki ako neko ima znanje da ga pogleda, ja sam glup :(

npr lokali koji ne rade (na kraju loga se zovu bas ti lokali jedan po jedan)
509, 510, 511, 512

[ eki_yu @ 20.07.2018. 06:58 ] @
Mozda lupam, ali ovo mi izgleda sumnjivo:

Code:

i sip_additional.conf
Code:


[446]
deny=0.0.0.0/0.0.0.0


Zar ne bi trebalo da bude allow za lokalne IP ?
[ bogdan.kecman @ 20.07.2018. 07:42 ] @
nemam pojma, ko sto rekoh ja sam za ovo polupismen, ali tako su namesteni svi lokali (tako ih je freepbx nasetovo taj fajl se autogenerise)... nemam ideju sta se tu deny uopste :( ali sam dobio kroz ovu pricu jednu ideju koju sad testiram pa cu javiti ako proradi

Code:

[[email protected] asterisk]# grep deny sip_additional.conf
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
[[email protected] asterisk]#

[ valjan @ 20.07.2018. 08:05 ] @
Uvek se stavi najpre deny za sve mreže, dakle za 0.0.0.0/0.0.0.0, a onda se stavi explicitni allow za lokalne mrežne opsege, znači u ovom slučaju bi to bilo 192.168.1.0/255.255.255.0, i to baš tim redom, i na tak način se Asterisk štiti od upada spolja.
[ bogdan.kecman @ 20.07.2018. 09:14 ] @
ne znam sta bi ti rekao, svuda ima SAMO deny, nema allow u celom fajlu...

inace pazi sad rad, ukacim se na telefon preko web-a kroz ssh tunel, promenim mu port na TCP, snimim, ne uspe da se registruje sto je ocekivano, promenim mu port nazad na UDP i sve radi ?!?!? i evo sad radi .. vec dva komada sam tako namestio i rade .. izgleda ovi provisioning confizi iz nekog razloga ne rade kako treba na ovim novim telefonima.. totalni zbun, isti telefoni, isti firmware
[ valjan @ 20.07.2018. 09:47 ] @
Ja sam imao problem sa jednim Cisco telefonom, jedva ga podesio preko provisioninga da se registruje na FreePBX, posle pola godine odlučili da mu promene lokal, kako smo izmenili broj lokala u provisioning fajlu, tako više nije mogao da se registruje šta god mu radio... E da, ne treba ti allow nego permit, i ako ga nema u sip_additional.conf ne znači da ga nema negde u sip.conf ili nekom drugom pomoćnom sip_nešto.conf fajlu, i ako se sva ova deny pravila parsiraju pre tog globalnog permit, ne bi trebalo da bude problema sa pristupom. Malko sam pomešao allow i disallow pravila za kodeke sa deny i permit pravilima za mrežni pristup, kada sve slično zvuči :-)
[ rajco @ 20.07.2018. 11:21 ] @
Uh, stavio si materijala za sat vremena gledanja :). Pošto imaš dosta telefona, moja ti je preporuka da koristiš endpoint manager za autoprovision, jednostavniji je, konforniji, sigurniji... Takođe ako su ti svi Yealink onda možeš dodatno da koristiš i RPS, super je kombinacija.
Hteo sam da ti pomenem i TCP dump da vidimo šta veli, ali reko i ovo će nam biti dosta...

@valjan ako ti treba za Cisco imam nekih template, tu je problem ako ti nije ok bukvalno jedan karakter neće proći provisioning.


[Ovu poruku je menjao rajco dana 20.07.2018. u 12:47 GMT+1]
[ bogdan.kecman @ 20.07.2018. 12:54 ] @
@valjan, ma ima milion cfg fajlova valjda kako koji deo app-a menja koji deo configa pa su razdelili a ja nemam ideju kako to normalno izgleda tako da .. sta mu ja mogu ... u svakom slucaju nije taj deo problem, kada sam usao na telefon preko veba i promenio setovanje i snimio oni rade tako da je problem do mojih cfg fajlova nesto im se tu ne svidja

sto se ciska tice ja sam imao tu 10tak cisko telefona duugo vremena nisam imao problema sa provisioningom uopste jedino sam morao da ih flashnem sa sip firmware-om jer originalni sa kojim dolazi ne zna sta je sip uopste

@rajco, probao sam te neke endpoint managere (2-3 razlicita) i iskreno ispostavilo se da mi je lakse da kreiram nove cfg fajlove rucno (evo tek posle preko godinu dana se pojavio problem prvi put), nisu svi yealink ... tcpdump sam gledao, ne pokazuje nista dodatno sto ne pokazuje asterisk log sa debug-om, problem je kad telefon kaze 404 ili kada rejectuje sa 503 ni u tcpdumpu ne pise nista vise a glupavi telefon ne kaze zasto je rejectovo... nisam gledao ovaj RPS servis ali mi deluje da je to neki komercijalni proizvod i da nesto tu sedi van lan-a, to otprilike ne dolazi u obzir


[ rajco @ 20.07.2018. 13:38 ] @
Pričam o endpoint manageru koji je sangomin, samo se plaća 75$, što se tiče RPS jeste Yealinkov proizvod i sedi van LAN-a, ali tebi ako telefoni imaju izlaz na internet u svakom slučaju kontaktiraju RPS, naravno ako nisi drop svog saobraćaja na njima osim LAN...
[ bogdan.kecman @ 20.07.2018. 13:41 ] @
nemaju telefoni izlaz na intrnet :D nijedan