[solved] Setting outgoing call dengan analog pabx

risa.futbas16 · April 13, 2017, 1:56am

Selamat Pagi,

Saya newbie untuk mencoba asterisk.
Saat ini saya ingin mengintegrasikan analog pabx, gateway grandstream gxw4108, dan briker
Kondisi yang saya pakai saat ini seperti ini :
PSTN telkom > analog pabx (extention 110) > grandstream gxw4108 (FX01) > Briker > Switch > User (ipphone)
Untuk telepon antar internal extention sudah bisa dilakukan, tetapi saya ada kendala dengan panggilan keluar (ke pstn lain atau mobile phone).
Jika ingin telepon keluar (pstn lain/mobilephone), di analog pabx disetting untuk menekan angka 9+nomor tujuan.
Saya sudah coba setting pada outbound Briker dengan 9|. ataupun 9X., tetapi masih belum bisa telepon keluar.
Bagaimana setting outgoing call ke pstn lain/mobile phone dengan kondisi yang saya inginkan?
Mohon untuk sharing info dan advicenya
Terima kasih

–
Regards
Risa

asoka · April 13, 2017, 3:26am

Halo,

Agar memudahkan dalam memberikan solusi, tolong dikirimkan konfigurasi yang sudah dilakukan pada Briker terutama bagian trunk dan outbound routes, serta pada Grandstream bagian trunk dan dialplan.

risa.futbas16 · April 13, 2017, 4:15am

Hallo,

Terlampir setting yang saya pakai saat ini baik di grandstream maupun asterisk.
Briker

outgoing

Selection_001.png481×608 82.5 KB
trunks

Selection_002.png430×593 35 KB

Grandstream
dialplan

Selection_004.png770×439 53.6 KB
profile1

Selection_003.png767×524 71.5 KB

Apa settingan saya masih ada yang kurang ya, sehingga tidak bisa telepon keluar pstn lain?
Mohon advicenya.
Terima kasih

asoka · April 13, 2017, 4:45am

Ok, coba pada bagian Briker dulu :

Pada konfigurasi trunk buat seperti ini :

host=192.168.1.193
context=from-trunk
type=peer
qualify=yes
nat=no
directmedia=no
insecure=port,invite
disallow=all
allow=alaw&ulaw
transport=udp

Pada bagian Outbound Routes cukup :

X.

Apabila sudah berhasil baru dimodify lagi outbound routesnya.

Setelah itu coba lakukan test call, jalankan perintah berikut sebelum melakukan test call :

asterisk -rx "sip set debug peer pbxtest"
tail -f /var/log/asterisk/full

Paste disini apabila masih error output log file full pada saat test call.

risa.futbas16 · April 13, 2017, 6:20am

Hallo,

Perintah tail -f /var/log/asterisk/full tidak bisa dilakukan pada briker, saya menggunakan perintah asterisks -rvvvvv.
Saya test telepon sesama internal extention pabx masih berfungsi normal, tetapi untuk telepon ke pstn lain atau mobile phone masih belum bisa dan saat melakukan panggilan ke lain pstn/mobile phone, tidak ada log yang tercreate
Berikut ini log saat saya coba ke extention analog pabx yang lain

Mohon advice nya kembali
terima kasih

asoka · April 13, 2017, 6:30am

Gpp pakai asterisk console juga, tapi kirimkan disini mulai dari proses callnya, jangan kirim sebagian seperti itu.

risa.futbas16 · April 13, 2017, 6:58am

Hallo,

saya kirim 2 part untuk lognya (karena terbatasnya character)

log part 1
root@ippbx:/home/support# asterisk -rvvvvv
Asterisk 13.9.1, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for detail s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
=========================================================================
Connected to Asterisk 13.9.1 currently running on ippbx (pid = 1371)
– Remote UNIX connection
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [112@from-internal:1] Macro(“SIP/1102-00000030”, “user-callerid,SKIPTTL,”) in new stack
– Executing [s@macro-user-callerid:1] NoOp(“SIP/1102-00000030”, “user-callerid: device 1102”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/1102-00000030”, “AMPUSER=1102”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/1102-00000030”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/1102-00000030”, “1?Set(REALCALLERIDNUM=1102)”) in new stack
– Executing [s@macro-user-callerid:5] NoOp(“SIP/1102-00000030”, “REALCALLERIDNUM is 1102”) in new stack
– Executing [s@macro-user-callerid:6] Set(“SIP/1102-00000030”, “AMPUSER=1102”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/1102-00000030”, “AMPUSERCIDNAME=IT JKT”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/1102-00000030”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/1102-00000030”, “AMPUSERCID=1102”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/1102-00000030”, “CALLERID(all)=“IT JKT” <1102>”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/1102-00000030”, “REALCALLERIDNUM=1102”) in new stack
– Executing [s@macro-user-callerid:12] ExecIf(“SIP/1102-00000030”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:13] NoOp(“SIP/1102-00000030”, “TTL: ARG1: SKIPTTL”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“SIP/1102-00000030”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [s@macro-user-callerid:23] NoOp(“SIP/1102-00000030”, “Using CallerID “IT JKT” <1102>”) in new stack
– Executing [112@from-internal:2] Set(“SIP/1102-00000030”, “_NODEST=”) in new stack
– Executing [112@from-internal:3] Macro(“SIP/1102-00000030”, “record-enable,1102,OUT,”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/1102-00000030”, “0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/1102-00000030”, “recordingcheck,20170413-134852,BrikerIPPBX-1492066132.92”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20170413-134852,BrikerIPPBX-1492066132.92: Outbound recording not enabled
– <SIP/1102-00000030>AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] NoOp(“SIP/1102-00000030”, “No recording needed”) in new stack
– Executing [112@from-internal:4] Macro(“SIP/1102-00000030”, “dialout-trunk,5,112,”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/1102-00000030”, “ENTEREDEXT=”) in new stack
– Executing [s@macro-dialout-trunk:2] NoOp(“SIP/1102-00000030”, “DEBUG: BT = CH = SIP/1102-00000030”) in new stack
– Executing [s@macro-dialout-trunk:3] Set(“SIP/1102-00000030”, “BT=”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/1102-00000030”, “BT=”) in new stack
– Executing [s@macro-dialout-trunk:5] NoOp(“SIP/1102-00000030”, “DEBUG: BT = CID = 1102”) in new stack
– Executing [s@macro-dialout-trunk:6] GotoIf(“SIP/1102-00000030”, “1?authkeylock”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] Set(“SIP/1102-00000030”, “KEYLOCK=”) in new stack
– Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/1102-00000030”, “0?askpin:pinok”) in new stack
– Goto (macro-dialout-trunk,s,13)
– Executing [s@macro-dialout-trunk:13] NoOp(“SIP/1102-00000030”, “OK”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/1102-00000030”, “DIAL_TRUNK=5”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“SIP/1102-00000030”, “ZAP2DAHDI=SIP/pbxtest”) in new stack
– Executing [s@macro-dialout-trunk:16] Set(“SIP/1102-00000030”, “VAR1=SIP/”) in new stack
– Executing [s@macro-dialout-trunk:17] Set(“SIP/1102-00000030”, “VAR2=pbxtest”) in new stack
– Executing [s@macro-dialout-trunk:18] ExecIf(“SIP/1102-00000030”, “0?Set(OUT_5=DAHDI/pbxtest)”) in new stack
– Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/1102-00000030”, “0?Set(OUT_5=DAHDI/pbxtest)”) in new stack
– Executing [s@macro-dialout-trunk:20] ExecIf(“SIP/1102-00000030”, “0?Set(OUT_5=DAHDI/pbxtest)”) in new stack
– Executing [s@macro-dialout-trunk:21] ExecIf(“SIP/1102-00000030”, “0?Authenticate()”) in new stack
– Executing [s@macro-dialout-trunk:22] GotoIf(“SIP/1102-00000030”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:23] Set(“SIP/1102-00000030”, “DIAL_NUMBER=112”) in new stack
– Executing [s@macro-dialout-trunk:24] Set(“SIP/1102-00000030”, “DIAL_TRUNK_OPTIONS=tTWr”) in new stack
– Executing [s@macro-dialout-trunk:25] Set(“SIP/1102-00000030”, “GROUP()=OUT_5”) in new stack
– Executing [s@macro-dialout-trunk:26] GotoIf(“SIP/1102-00000030”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,28)
– Executing [s@macro-dialout-trunk:28] GotoIf(“SIP/1102-00000030”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:29] Set(“SIP/1102-00000030”, “DIAL_TRUNK_OPTIONS=TW”) in new stack
– Executing [s@macro-dialout-trunk:30] Macro(“SIP/1102-00000030”, “outbound-callerid,5”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/1102-00000030”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] GotoIf(“SIP/1102-00000030”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,4)
– Executing [s@macro-outbound-callerid:4] NoOp(“SIP/1102-00000030”, “REALCALLERIDNUM is 1102”) in new stack
– Executing [s@macro-outbound-callerid:5] GotoIf(“SIP/1102-00000030”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,10)
– Executing [s@macro-outbound-callerid:10] Set(“SIP/1102-00000030”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:11] Set(“SIP/1102-00000030”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:12] Set(“SIP/1102-00000030”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:13] GotoIf(“SIP/1102-00000030”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,17)
– Executing [s@macro-outbound-callerid:17] GotoIf(“SIP/1102-00000030”, “1?usercid”) in new stack
– Goto (macro-outbound-callerid,s,19)
– Executing [s@macro-outbound-callerid:19] GotoIf(“SIP/1102-00000030”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,23)
– Executing [s@macro-outbound-callerid:23] NoOp(“SIP/1102-00000030”, “CallerID set to “IT JKT” <1102>”) in new stack
– Executing [s@macro-dialout-trunk:31] AGI(“SIP/1102-00000030”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
– <SIP/1102-00000030>AGI Script fixlocalprefix completed, returning 0
– Executing [s@macro-dialout-trunk:32] Set(“SIP/1102-00000030”, “OUTNUM=112”) in new stack
– Executing [s@macro-dialout-trunk:33] Set(“SIP/1102-00000030”, “REALOUTNUM=112”) in new stack
– Executing [s@macro-dialout-trunk:34] GotoIf(“SIP/1102-00000030”, “0?askpin1:pinok1”) in new stack
– Goto (macro-dialout-trunk,s,39)
– Executing [s@macro-dialout-trunk:39] NoOp(“SIP/1102-00000030”, “OK”) in new stack
– Executing [s@macro-dialout-trunk:40] Set(“SIP/1102-00000030”, “OUTNUM=112”) in new stack
– Executing [s@macro-dialout-trunk:41] Set(“SIP/1102-00000030”, “DEVICENUM=1102”) in new stack
– Executing [s@macro-dialout-trunk:42] Set(“SIP/1102-00000030”, “CIDNAME=”) in new stack
– Executing [s@macro-dialout-trunk:43] Set(“SIP/1102-00000030”, “CIDNUM=”) in new stack
– Executing [s@macro-dialout-trunk:44] ExecIf(“SIP/1102-00000030”, “1?Set(CIDNUM=1102)”) in new stack
– Executing [s@macro-dialout-trunk:45] Set(“SIP/1102-00000030”, “CDR(accountcode)=1102”) in new stack
– Executing [s@macro-dialout-trunk:46] GotoIf(“SIP/1102-00000030”, “0?ACCOUNTCODECFOK”) in new stack
– Executing [s@macro-dialout-trunk:47] ExecIf(“SIP/1102-00000030”, “0?Set(CDR(accountcode)=1102)”) in new stack
– Executing [s@macro-dialout-trunk:48] Goto(“SIP/1102-00000030”, “ACCOUNTCODECFNOTOK”) in new stack
– Goto (macro-dialout-trunk,s,54)
– Executing [s@macro-dialout-trunk:54] Set(“SIP/1102-00000030”, “custom=SIP/pbxtest”) in new stack
– Executing [s@macro-dialout-trunk:55] GotoIf(“SIP/1102-00000030”, “1?gocall”) in new stack
– Goto (macro-dialout-trunk,s,57)
– Executing [s@macro-dialout-trunk:57] Macro(“SIP/1102-00000030”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk:58] GotoIf(“SIP/1102-00000030”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:59] GotoIf(“SIP/1102-00000030”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:60] Dial(“SIP/1102-00000030”, “SIP/pbxtest/112,300,TW”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 12926
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.193:5060:
INVITE sip:112@192.168.1.193 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
Max-Forwards: 70
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193
Contact: sip:1102@192.168.1.219:5060
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 102 INVITE
User-Agent: BrikerIPPBX
Date: Thu, 13 Apr 2017 06:48:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 231893146 231893146 IN IP4 192.168.1.219
s=Asterisk PBX 13.9.1
c=IN IP4 192.168.1.219
t=0 0
m=audio 12926 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

-- Called SIP/pbxtest/112

<— SIP read from UDP:192.168.1.193:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 102 INVITE
User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
Content-Length: 0

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

<— SIP read from UDP:192.168.1.193:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193;tag=fc50730023716f24
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 102 INVITE
User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
Contact: sip:192.168.1.193:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

<------------->
— (10 headers 0 lines) —
sip_route_dump: route/path hop: sip:192.168.1.193:5060;transport=udp
– SIP/pbxtest-00000031 is ringing

<— SIP read from UDP:192.168.1.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193;tag=fc50730023716f24
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 102 INVITE
User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
Contact: sip:192.168.1.193:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Supported: replaces, timer, 100rel, path
Content-Length: 224

v=0
o=system 8006 8000 IN IP4 192.168.1.193
s=SIP Call
c=IN IP4 192.168.1.193
t=0 0
m=audio 5028 RTP/AVP 8 4 18 3 0 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
— (12 headers 11 lines) —
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|gsm|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.193:5028
Peer doesn’t provide T.140
sip_route_dump: route/path hop: sip:192.168.1.193:5060;transport=udp
set_destination: Parsing sip:192.168.1.193:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.193:5060
Transmitting (no NAT) to 192.168.1.193:5060:
ACK sip:192.168.1.193:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK5ca766a7
Max-Forwards: 70
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193;tag=fc50730023716f24
Contact: sip:1102@192.168.1.219:5060
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 102 ACK
User-Agent: BrikerIPPBX
Content-Length: 0

<— SIP read from UDP:192.168.1.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193;tag=fc50730023716f24
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 102 INVITE
User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
Contact: sip:192.168.1.193:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Supported: replaces, timer, 100rel, path
Content-Length: 224

v=0
o=system 8006 8001 IN IP4 192.168.1.193
s=SIP Call
c=IN IP4 192.168.1.193
t=0 0
m=audio 5028 RTP/AVP 8 4 18 3 0 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
— (12 headers 11 lines) —
– SIP/pbxtest-00000031 answered SIP/1102-00000030
set_destination: Parsing sip:192.168.1.193:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.193:5060
Transmitting (no NAT) to 192.168.1.193:5060:
ACK sip:192.168.1.193:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK0e4bf795
Max-Forwards: 70
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193;tag=fc50730023716f24
Contact: sip:1102@192.168.1.219:5060
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 102 ACK
User-Agent: BrikerIPPBX
Content-Length: 0

risa.futbas16 · April 13, 2017, 6:59am

Hallo,

log part 2

-- Channel SIP/pbxtest-00000031 joined 'simple_bridge' basic-bridge <e4085bc0-a382-4a54-9c24-f2840be99507>
-- Channel SIP/1102-00000030 joined 'simple_bridge' basic-bridge <e4085bc0-a382-4a54-9c24-f2840be99507>
   > 0x7f9b443a4ec0 -- Probation passed - setting RTP source address to 192.168.1.193:5028
   > 0x7f9b480418c0 -- Probation passed - setting RTP source address to 192.168.1.133:10010
-- Channel SIP/1102-00000030 left 'simple_bridge' basic-bridge <e4085bc0-a382-4a54-9c24-f2840be99507>
-- Channel SIP/pbxtest-00000031 left 'simple_bridge' basic-bridge <e4085bc0-a382-4a54-9c24-f2840be99507>

Scheduling destruction of SIP dialog ‘7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:192.168.1.193:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.193:5060
Reliably Transmitting (no NAT) to 192.168.1.193:5060:
BYE sip:192.168.1.193:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK5440e753
Max-Forwards: 70
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193;tag=fc50730023716f24
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 103 BYE
User-Agent: BrikerIPPBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (macro-dialout-trunk, s, 60) exited non-zero on ‘SIP/1102-00000030’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 112, 4) exited non-zero on ‘SIP/1102-00000030’
– Executing [h@from-internal:1] Macro(“SIP/1102-00000030”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/1102-00000030”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/1102-00000030”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/1102-00000030”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“SIP/1102-00000030”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/1102-00000030’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1102-00000030’

<— SIP read from UDP:192.168.1.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK5440e753
From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
To: sip:112@192.168.1.193;tag=fc50730023716f24
Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
CSeq: 103 BYE
User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
Contact: sip:192.168.1.193:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer, 100rel, path
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060’ Method: INVITE
Reliably Transmitting (no NAT) to 192.168.1.193:5060:
OPTIONS sip:192.168.1.193 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK00e92a00
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.1.219;tag=as3eeaa749
To: sip:192.168.1.193
Contact: sip:Unknown@192.168.1.219:5060
Call-ID: 3464a8c044f067c75a19e3da26faa298@192.168.1.219:5060
CSeq: 102 OPTIONS
User-Agent: BrikerIPPBX
Date: Thu, 13 Apr 2017 06:49:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK00e92a00
From: “Unknown” sip:Unknown@192.168.1.219;tag=as3eeaa749
To: sip:192.168.1.193;tag=17e2fe4225a6aac4
Call-ID: 3464a8c044f067c75a19e3da26faa298@192.168.1.219:5060
CSeq: 102 OPTIONS
User-Agent: Grandstream GXW4108 (HW 1.1, Ch:8) 1.3.4.13
Contact: sip:192.168.1.193:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer, 100rel, path
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘3464a8c044f067c75a19e3da26faa298@192.168.1.219:5060’ Method: OPTIONS
ippbxCLI> ippbxCLI> User-Agent: Grandstream GXW4108 (HW 1.1, Ch:5) 1.3.4.13
ippbxCLI> 0;traippbxCLI> Contact: sip:192.168.1.193:5060;transport=udp
No such command ‘ippbxCLI> User-Agent: Grandstream GXW4108 (HW 1.1, Ch:5) 1.3.4.13’ (type 'core show help ippbxCLI> User-Agent:’ for other possib
No such command ‘0;traippbxCLI> Contact: sip:192.168.1.193:5060;transport=udp’ (type 'core show help 0;traippbxCLI> Contact:’ for other possibl
ippbxCLI> root@ippbx:/home/support# asterisk -rvvvvv
ippbxCLI> Asterisk 13.9.1, Copyright © 1999 - 2014, Digium, Inc. and others.
ippbxCLI> Created by Mark Spencer markster@digium.com
ippbxCLI> Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for detail s.
ippbxCLI> This is free software, with components licensed under the GNU General Public
ippbxCLI> License version 2 and other licenses; you are welcome to redistribute it under
ippbxCLI> certain conditions. Type ‘core show license’ for details.
ippbxCLI> =========================================================================
ippbxCLI> Connected to Asterisk 13.9.1 currently running on ippbx (pid = 1371)
alout-ippbxCLI> – Remote UNIX connection
ippbxCLI> == Using SIP VIDEO TOS bits 136
ippbxCLI> == Using SIP VIDEO CoS mark 6
ippbxCLI> == Using SIP RTP TOS bits 184
ippbxCLI> == Using SIP RTP CoS mark 5
ippbxCLI> – Executing [112@from-internal:1] Macro(“SIP/1102-00000030”, “user-callerid,SKIPTTL,”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:1] NoOp(“SIP/1102-00000030”, “user-callerid: device 1102”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:2] Set(“SIP/1102-00000030”, “AMPUSER=1102”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:3] GotoIf(“SIP/1102-00000030”, “0report”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:4] ExecIf(“SIP/1102-00000030”, “1Set(REALCALLERIDNUM=1102)”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:5] NoOp(“SIP/1102-00000030”, “REALCALLERIDNUM is 1102”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:6] Set(“SIP/1102-00000030”, “AMPUSER=1102”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:7] Set(“SIP/1102-00000030”, “AMPUSERCIDNAME=IT JKT”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:8] GotoIf(“SIP/1102-00000030”, “0report”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:9] Set(“SIP/1102-00000030”, “AMPUSERCID=1102”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:10] Set(“SIP/1102-00000030”, “CALLERID(all)=“IT JKT” <1102>”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:11] Set(“SIP/1102-00000030”, “REALCALLERIDNUM=1102”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:12] ExecIf(“SIP/1102-00000030”, “0Set(CHANNEL(language)=)”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:13] NoOp(“SIP/1102-00000030”, “TTL: ARG1: SKIPTTL”) in new stack
ippbxCLI> – Executing [s@macro-user-callerid:14] GotoIf(“SIP/1102-00000030”, “1continue”) in new stack
ippbxCLI> – Goto (macro-user-callerid,s,23)
ippbxCLI> – Executing [s@macro-user-callerid:23] NoOp(“SIP/1102-00000030”, “Using CallerID “IT JKT” <1102>”) in new stack
ippbxCLI> – Executing [112@from-internal:2] Set(“SIP/1102-00000030”, “_NODEST=”) in new stack
ippbxCLI> – Executing [112@from-internal:3] Macro(“SIP/1102-00000030”, “record-enable,1102,OUT,”) in new stack
ippbxCLI> – Executing [s@macro-record-enable:1] GotoIf(“SIP/1102-00000030”, “02:4”) in new stack
ippbxCLI> – Goto (macro-record-enable,s,4)
ippbxCLI> – Executing [s@macro-record-enable:4] AGI(“SIP/1102-00000030”, “recordingcheck,20170413-134852,BrikerIPPBX-1492066132.92”) in new s
ippbxCLI> – Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
ippbxCLI> recordingcheck,20170413-134852,BrikerIPPBX-1492066132.92: Outbound recording not enabled
ippbxCLI> – <SIP/1102-00000030>AGI Script recordingcheck completed, returning 0
ippbxCLI> – Executing [s@macro-record-enable:5] NoOp(“SIP/1102-00000030”, “No recording needed”) in new stack
ippbxCLI> – Executing [112@from-internal:4] Macro(“SIP/1102-00000030”, “dialout-trunk,5,112,”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:1] Set(“SIP/1102-00000030”, “ENTEREDEXT=”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:2] NoOp(“SIP/1102-00000030”, “DEBUG: BT = CH = SIP/1102-00000030”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:3] Set(“SIP/1102-00000030”, “BT=”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:4] Set(“SIP/1102-00000030”, “BT=”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:5] NoOp(“SIP/1102-00000030”, “DEBUG: BT = CID = 1102”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:6] GotoIf(“SIP/1102-00000030”, “1authkeylock”) in new stack
ippbxCLI> – Goto (macro-dialout-trunk,s,9)
ippbxCLI> – Executing [s@macro-dialout-trunk:9] Set(“SIP/1102-00000030”, “KEYLOCK=”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/1102-00000030”, “0askpin:pinok”) in new stack
ippbxCLI> – Goto (macro-dialout-trunk,s,13)
ippbxCLI> – Executing [s@macro-dialout-trunk:13] NoOp(“SIP/1102-00000030”, “OK”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:14] Set(“SIP/1102-00000030”, “DIAL_TRUNK=5”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:15] Set(“SIP/1102-00000030”, “ZAP2DAHDI=SIP/pbxtest”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:16] Set(“SIP/1102-00000030”, “VAR1=SIP/”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:17] Set(“SIP/1102-00000030”, “VAR2=pbxtest”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:18] ExecIf(“SIP/1102-00000030”, “0Set(OUT_5=DAHDI/pbxtest)”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/1102-00000030”, “0Set(OUT_5=DAHDI/pbxtest)”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:20] ExecIf(“SIP/1102-00000030”, “0Set(OUT_5=DAHDI/pbxtest)”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:21] ExecIf(“SIP/1102-00000030”, “0Authenticate()”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:22] GotoIf(“SIP/1102-00000030”, “0disabletrunk,1”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:23] Set(“SIP/1102-00000030”, “DIAL_NUMBER=112”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:24] Set(“SIP/1102-00000030”, “DIAL_TRUNK_OPTIONS=tTWr”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:25] Set(“SIP/1102-00000030”, “GROUP()=OUT_5”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:26] GotoIf(“SIP/1102-00000030”, “1nomax”) in new stack
ippbxCLI> – Goto (macro-dialout-trunk,s,28)
ippbxCLI> – Executing [s@macro-dialout-trunk:28] GotoIf(“SIP/1102-00000030”, “0skipoutcid”) in new stack
ng (no NAT) ippbxCLI> – Executing [s@macro-dialout-trunk:29] Set(“SIP/1102-00000030”, “DIAL_TRUNK_OPTIONS=TW”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:30] Macro(“SIP/1102-00000030”, “outbound-callerid,5”) in new stack
.168.1.ippbxCLI> – Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/1102-00000030”, “0Set(CALLERPRES()=)”) in new stack
Seq: 1ippbxCLI> – Executing [s@macro-outbound-callerid:2] GotoIf(“SIP/1102-00000030”, “1start”) in new stack
ippbxCLI> – Goto (macro-outbound-callerid,s,4)
ippbxCLI> – Executing [s@macro-outbound-callerid:4] NoOp(“SIP/1102-00000030”, “REALCALLERIDNUM is 1102”) in new stack
ippbxCLI> – Executing [s@macro-outbound-callerid:5] GotoIf(“SIP/1102-00000030”, “1normcid”) in new stack
ippbxCLI> – Goto (macro-outbound-callerid,s,10)
ippbxCLI> – Executing [s@macro-outbound-callerid:10] Set(“SIP/1102-00000030”, “USEROUTCID=”) in new stack
ippbxCLI> – Executing [s@macro-outbound-callerid:11] Set(“SIP/1102-00000030”, “EMERGENCYCID=”) in new stack
ippbxCLI> – Executing [s@macro-outbound-callerid:12] Set(“SIP/1102-00000030”, “TRUNKOUTCID=”) in new stack
ippbxCLI> – Executing [s@macro-outbound-callerid:13] GotoIf(“SIP/1102-00000030”, “1trunkcid”) in new stack
ippbxCLI> – Goto (macro-outbound-callerid,s,17)
ippbxCLI> – Executing [s@macro-outbound-callerid:17] GotoIf(“SIP/1102-00000030”, “1usercid”) in new stack
ippbxCLI> – Goto (macro-outbound-callerid,s,19)
180 RingippbxCLI> – Executing [s@macro-outbound-callerid:19] GotoIf(“SIP/1102-00000030”, “1report”) in new stack
68.1.219ippbxCLI> – Goto (macro-outbound-callerid,s,23)
ippbxCLI> – Executing [s@macro-outbound-callerid:23] NoOp(“SIP/1102-00000030”, “CallerID set to “IT JKT” <1102>”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:31] AGI(“SIP/1102-00000030”, “fixlocalprefix”) in new stack
ippbxCLI> – Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
ippbxCLI> – <SIP/1102-00000030>AGI Script fixlocalprefix completed, returning 0
ippbxCLI> – Executing [s@macro-dialout-trunk:32] Set(“SIP/1102-00000030”, “OUTNUM=112”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:33] Set(“SIP/1102-00000030”, “REALOUTNUM=112”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:34] GotoIf(“SIP/1102-00000030”, “0askpin1:pinok1”) in new stack
ippbxCLI> – Goto (macro-dialout-trunk,s,39)
ippbxCLI> – Executing [s@macro-dialout-trunk:39] NoOp(“SIP/1102-00000030”, “OK”) in new stack
108 (ippbxCLI> – Executing [s@macro-dialout-trunk:40] Set(“SIP/1102-00000030”, “OUTNUM=112”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:41] Set(“SIP/1102-00000030”, “DEVICENUM=1102”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:42] Set(“SIP/1102-00000030”, “CIDNAME=”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:43] Set(“SIP/1102-00000030”, “CIDNUM=”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:44] ExecIf(“SIP/1102-00000030”, “1Set(CIDNUM=1102)”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:45] Set(“SIP/1102-00000030”, “CDR(accountcode)=1102”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:46] GotoIf(“SIP/1102-00000030”, “0ACCOUNTCODECFOK”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:47] ExecIf(“SIP/1102-00000030”, “0Set(CDR(accountcode)=1102)”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:48] Goto(“SIP/1102-00000030”, “ACCOUNTCODECFNOTOK”) in new stack
ippbxCLI> – Goto (macro-dialout-trunk,s,54)
lephone-ippbxCLI> – Executing [s@macro-dialout-trunk:54] Set(“SIP/1102-00000030”, “custom=SIP/pbxtest”) in new stack
oippbxCLI> – Executing [s@macro-dialout-trunk:55] GotoIf(“SIP/1102-00000030”, “1gocall”) in new stack
ippbxCLI> – Goto (macro-dialout-trunk,s,57)
ippbxCLI> – Executing [s@macro-dialout-trunk:57] Macro(“SIP/1102-00000030”, “dialout-trunk-predial-hook,”) in new stack
) ippbxCLI> – Executing [s@macro-dialout-trunk:58] GotoIf(“SIP/1102-00000030”, “0bypass,1”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:59] GotoIf(“SIP/1102-00000030”, “0customtrunk”) in new stack
ippbxCLI> – Executing [s@macro-dialout-trunk:60] Dial(“SIP/1102-00000030”, “SIP/pbxtest/112,300,TW”) in new stack
ippbxCLI> == Using SIP RTP TOS bits 184
ippbxCLI> == Using SIP RTP CoS mark 5
User-AippbxCLI> Audio is at 12926
gent: BrikerIPPBX
ippbxCLI> Adding codec alaw to SDP
ippbxCLI> Adding codec ulaw to SDP
ippbxCLI> Adding non-codec 0x1 (telephone-event) to SDP
ippbxCLI> Reliably Transmitting (no NAT) to 192.168.1.193:5060:
ippbxCLI> INVITE sip:112@192.168.1.193 SIP/2.0
ippbxCLI> Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
ippbxCLI> Max-Forwards: 70
ippbxCLI> From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
ippbxCLI> To: sip:112@192.168.1.193
ippbxCLI> Contact: sip:1102@192.168.1.219:5060
ippbxCLI> Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
ippbxCLI> CSeq: 102 INVITE
ippbxCLI> User-Agent: BrikerIPPBX
ippbxCLI> Date: Thu, 13 Apr 2017 06:48:53 GMT
nt-Type:ippbxCLI> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
ippbxCLI> Supported: replaces, timer
ippbxCLI> Content-Type: application/sdp
ippbxCLI> Content-Length: 263
68.1.193
t=0 0
m=audippbxCLI>
ippbxCLI> v=0
ippbxCLI> o=root 231893146 231893146 IN IP4 192.168.1.219
ippbxCLI> s=Asterisk PBX 13.9.1
ippbxCLI> c=IN IP4 192.168.1.219
ippbxCLI> t=0 0
ippbxCLI> m=audio 12926 RTP/AVP 8 0 101
ippbxCLI> a=rtpmap:8 PCMA/8000
ippbxCLI> a=rtpmap:0 PCMU/8000
ippbxCLI> a=rtpmap:101 telephone-event/8000
ippbxCLI> a=fmtp:101 0-16
ippbxCLI> a=maxptime:150
ippbxCLI> a=sendrecv
/1102-00000030
set_destination: Parsing <sippbxCLI>
ippbxCLI> —
ippbxCLI> – Called SIP/pbxtest/112
ippbxCLI>
ippbxCLI> <— SIP read from UDP:192.168.1.193:5060 —>
tination: sippbxCLI> SIP/2.0 100 Trying
et destination to 1ippbxCLI> Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
ippbxCLI> From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
ippbxCLI> To: sip:112@192.168.1.193
ippbxCLI> Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
ippbxCLI> CSeq: 102 INVITE
ippbxCLI> <sip:1102@192.168User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
ippbxCLI> Content-Length: 0
c50730023716f24
CoippbxCLI>
ippbxCLI> <------------->
ippbxCLI> — (8 headers 0 lines) —
ippbxCLI>
ippbxCLI> <— SIP read from UDP:192.168.1.193:5060 —>
ippbxCLI> SIP/2.0 180 Ringing
5060
CSeq: 102 ACK
UippbxCLI> Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
ippbxCLI> From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
ippbxCLI> To: sip:112@192.168.1.193;tag=fc50730023716f24
ippbxCLI> Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
ippbxCLI> CSeq: 102 INVITE
ippbxCLI> User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
ippbxCLI> Contact: sip:192.168.1.193:5060;transport=udp
ippbxCLI> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
ippbxCLI> Content-Length: 0
ce address to 192.ippbxCLI>
ippbxCLI> <------------->
ippbxCLI> — (10 headers 0 lines) —
ippbxCLI> sip_route_dump: route/path hop: sip:192.168.1.193:5060;transport=udp
9ippbxCLI> – SIP/pbxtest-00000031 is ringing
ippbxCLI>
ippbxCLI> <— SIP read from UDP:192.168.1.193:5060 —>
ippbxCLI> SIP/2.0 200 OK
ippbxCLI> Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
d479a211ippbxCLI> From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
ippbxCLI> To: sip:112@192.168.1.193;tag=fc50730023716f24
ippbxCLI> Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
ippbxCLI> CSeq: 102 INVITE
et_destination: sippbxCLI> User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
ippbxCLI> Contact: sip:192.168.1.193:5060;transport=udp
ippbxCLI> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
ippbxCLI> Content-Type: application/sdp
ippbxCLI> Supported: replaces, timer, 100rel, path
ippbxCLI> Content-Length: 224
ippbxCLI>
o:ippbxCLI> v=0
ippbxCLI> o=system 8006 8000 IN IP4 192.168.1.193
ippbxCLI> s=SIP Call
ippbxCLI> -ID: c=IN IP4 192.168.1.193
ippbxCLI> t=0 0
ippbxCLI> m=audio 5028 RTP/AVP 8 4 18 3 0 101
ippbxCLI> a=sendrecv
ippbxCLI> a=rtpmap:8 PCMA/8000
ippbxCLI> a=ptime:20
ippbxCLI> a=rtpmap:101 telephone-event/8000
ippbxCLI> a=fmtp:101 0-11
ippbxCLI> <------------->
ippbxCLI> — (12 headers 11 lines) —
ippbxCLI> Found RTP audio format 8
ippbxCLI> Found RTP audio format 4
ippbxCLI> Found RTP audio format 18
ippbxCLI> macFound RTP audio format 3
ippbxCLI> Found RTP audio format 0
ippbxCLI> Found RTP audio format 101
ippbxCLI> non-zeFound audio description format PCMA for ID 8
ippbxCLI> Found audio description format telephone-event for ID 101
ippbxCLI> Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|gsm|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
ippbxCLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
ippbxCLI> Peer audio RTP is at port 192.168.1.193:5028
ippbxCLI> Peer doesn’t provide T.140
ippbxCLI> sip_route_dump: route/path hop: sip:192.168.1.193:5060;transport=udp
lippbxCLI> set_destination: Parsing sip:192.168.1.193:5060;transport=udp for address/port to send to
ippbxCLI> set_destination: set destination to 192.168.1.193:5060
ippbxCLI> Transmitting (no NAT) to 192.168.1.193:5060:
ippbxCLI> ACK sip:192.168.1.193:5060;transport=udp SIP/2.0
ippbxCLI> Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK5ca766a7
ippbxCLI> Max-Forwards: 70
ippbxCLI> From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
ippbxCLI> To: sip:112@192.168.1.193;tag=fc50730023716f24
ippbxCLI> Contact: sip:1102@192.168.1.219:5060
ippbxCLI> Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
ippbxCLI> CSeq: 102 ACK
ippbxCLI> User-Agent: BrikerIPPBX
ippbxCLI> Content-Length: 0
ippbxCLI>
ippbxCLI>
ippbxCLI> —
ippbxCLI>
sippbxCLI> <— SIP read from UDP:192.168.1.193:5060 —>
ippbxCLI> SIP/2.0 200 OK
ippbxCLI> Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK047aaa43
ippbxCLI> From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
ippbxCLI> To: sip:112@192.168.1.193;tag=fc50730023716f24
ippbxCLI> Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
ippbxCLI> CSeq: 102 INVITE
ippbxCLI> User-Agent: Grandstream GXW4108 (HW 1.1, Ch:6) 1.3.4.13
168.1.2ippbxCLI> Contact: sip:192.168.1.193:5060;transport=udp
ippbxCLI> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
ippbxCLI> Content-Type: application/sdp
ippbxCLI> Supported: replaces, timer, 100rel, path
ippbxCLI> Content-Length: 224
ippbxCLI>
ippbxCLI> v=0
ippbxCLI> o=system 8006 8001 IN IP4 192.168.1.193
ippbxCLI> s=SIP Call
ippbxCLI> c=IN IP4 192.168.1.193
ippbxCLI> t=0 0
1ippbxCLI> m=audio 5028 RTP/AVP 8 4 18 3 0 101
ippbxCLI> a=sendrecv
ippbxCLI> a=rtpmap:8 PCMA/8000
ippbxCLI> a=ptime:20
q: 1ippbxCLI> a=rtpmap:101 telephone-event/8000
ippbxCLI> a=fmtp:101 0-11
ippbxCLI> <------------->
ippbxCLI> — (12 headers 11 lines) —
ippbxCLI> – SIP/pbxtest-00000031 answered SIP/1102-00000030
ippbxCLI> set_destination: Parsing sip:192.168.1.193:5060;transport=udp for address/port to send to
:192.16ippbxCLI> set_destination: set destination to 192.168.1.193:5060
ippbxCLI> Transmitting (no NAT) to 192.168.1.193:5060:
ippbxCLI> ACK sip:192.168.1.193:5060;transport=udp SIP/2.0
ippbxCLI> Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK0e4bf795
D: 346ippbxCLI> Max-Forwards: 70
ippbxCLI> From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
ippbxCLI> To: sip:112@192.168.1.193;tag=fc50730023716f24
ippbxCLI> Contact: sip:1102@192.168.1.219:5060
ippbxCLI> Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
TIONS,ippbxCLI> CSeq: 102 ACK
ippbxCLI> User-Agent: BrikerIPPBX
ippbxCLI> Content-Length: 0
ippbxCLI>
ippbxCLI>
ippbxCLI> —
ippbxCLI> – Channel SIP/pbxtest-00000031 joined ‘simple_bridge’ basic-bridge <e4085bc0-a382-4a54-9c24-f284
ippbxCLI> – Channel SIP/1102-00000030 joined ‘simple_bridge’ basic-bridge <e4085bc0-a382-4a54-9c24-f
ippbxCLI> > 0x7f9b443a4ec0 – Probation passed - setting RTP source address to 192.168.1.19
ippbxCLI> > 0x7f9b480418c0 – Probation passed - setting RTP source address to 192.168.
ippbxCLI> – Channel SIP/1102-00000030 left ‘simple_bridge’ basic-bridge <e4085bc0-a382
ippbxCLI> – Channel SIP/pbxtest-00000031 left ‘simple_bridge’ basic-bridge <e408
ippbxCLI> Scheduling destruction of SIP dialog '7d479a2118297cab1ad172a737a28f9
ippbxCLI> set_destination: Parsing sip:192.168.1.193:5060;transport=udp for a
ippbxCLI> set_destination: set destination to 192.168.1.193:5060
ippbxCLI> Reliably Transmitting (no NAT) to 192.168.1.193:5060:
ippbxCLI> BYE sip:192.168.1.193:5060;transport=udp SIP/2.0
ippbxCLI> Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK5440e753
ippbxCLI> Max-Forwards: 70
ippbxCLI> From: “IT JKT” sip:1102@192.168.1.219;tag=as1655265b
ippbxCLI> To: sip:112@192.168.1.193;tag=fc50730023716f24
ippbxCLI> Call-ID: 7d479a2118297cab1ad172a737a28f98@192.168.1.219:5060
ippbxCLI> CSeq: 103 BYE
ippbxCLI> User-Agent: BrikerIPPBX
ippbxCLI> X-Asterisk-HangupCause: Normal Clearing
ippbxCLI> X-Asterisk-HangupCauseCode: 16
ippbxCLI> Content-Length: 0
ippbxCLI>
ippbxCLI>
ippbxCLI> —
ippbxCLI> == Spawn extension (macro-dialout-trunk, s, 60) exited non-zero on
ippbxCLI> == Spawn extension (from-internal, 112, 4) exited non-zero on 'SIP/
ippbxCLI> – Executing [h@from-internal:1] Macro(“SIP/1102-00000030”, "hang
ippbxCLI> – Executing [s@macro-hangupcall:1] GotoIf(“SIP/1102-00000030”, "
ippbxCLI> – Goto (macro-hangupcall,s,4)
ippbx*CLI> – Executing [s@macro-hangupcall:4] GotoIf(“SIP/1102-00000030”, "

Mohon advicenya kembali
Terima kasih

asoka · April 13, 2017, 7:33am

Kalo dari log tujuan 112 itu sudah ringing, apakah ditujuan tidak ringing atau bagaimana ?

risa.futbas16 · April 13, 2017, 7:44am

Hallo,

Untuk sesama user pabx yang extentionnya terdapat pada analog pabx, setting saat ini sudah bisa digunakan melalui ipphone.
Saat ini saya ada kendala jika saya telepon ke luar (PSTN lain atau mobile phone) dari ipphone.
Jika menggunakan telepon analog biasa, saya harus menekan angka 9+nomor tujuan, baru bisa telepon ke luar (PSTN lain atau mobile phone).
Mungkin saya masih ada salah pada setting grandstream atau asterisknya, sehingga saya hanya bisa incoming dan outgoing di internal extention saja dari ipphone.
Apakah ada pengalaman mengintegrasikan analog pabx, grandstream, dan asterisk?

–
Regards
Risa

asoka · April 13, 2017, 7:54am

Apabila sudah bisa ke extension di PBX Analog, berarti konfigurasi Trunk sudah OK.

Selanjutnya untuk call ke Nomor PSTN, coba di paste disini log asterisk pada saat call ke nomor PSTN seperti sebelumnya, gunakan pastebin.com apabila kesulitan dan kirim url nya disini.

risa.futbas16 · April 13, 2017, 8:11am

hallo,

Saya coba telepon dengan menekan 9+nomor tujuan di ipphone saya, hanya tertulis please hangup pada ipphone dan tidak ada log yang tercreate pada briker.
Begitu juga jika saya menekan nomor tujuan, log juga tidak tercreate pada briker tetapi telepon langsung terhubung ke extention operator kantor.
Tapi jika saya lakukan incoming ataupun outgoing call ke sesama user extention internal pabx muncul lognya (seperti yang saya kirimkan sebelumnya)
Mohon bantuannya kembali.

asoka · April 13, 2017, 8:20am

Ok, coba edit file /etc/asterisk/logger.conf

lalu hilangkan tanda ; pada ;full => notice,warning,error,debug,verbose

setelah itu jalankan perintah : asterisk -rx "logger reload"

Lalu jalankan perintah : tail -f /var/log/asterisk/full

Setelah itu test call ke nomor PSTN kembali, lalu perhatikan output log full tersebut.

risa.futbas16 · April 13, 2017, 8:26am

Hallo,

Terlampir log saat melakukan panggilan ke luar (pencet angka 9+nomor tujuan)
root@ippbx:/home/support# asterisk -rvvvvv
Asterisk 13.9.1, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for detail s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 13.9.1 currently running on ippbx (pid = 1371)
Reliably Transmitting (no NAT) to 192.168.1.193:5060:
OPTIONS sip:192.168.1.193 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK61c11010
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.1.219;tag=as4246ed4d
To: sip:192.168.1.193
Contact: sip:Unknown@192.168.1.219:5060
Call-ID: 4f8a137f38e47d5d3155121c4bc6de26@192.168.1.219:5060
CSeq: 102 OPTIONS
User-Agent: BrikerIPPBX
Date: Thu, 13 Apr 2017 08:24:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK61c11010
From: “Unknown” sip:Unknown@192.168.1.219;tag=as4246ed4d
To: sip:192.168.1.193;tag=4ba64fd6ab87ee60
Call-ID: 4f8a137f38e47d5d3155121c4bc6de26@192.168.1.219:5060
CSeq: 102 OPTIONS
User-Agent: Grandstream GXW4108 (HW 1.1, Ch:8) 1.3.4.13
Contact: sip:5060@192.168.1.193:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer, 100rel, path
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘4f8a137f38e47d5d3155121c4bc6de26@192.168.1.219:506 0’ Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.1.193:5060:
OPTIONS sip:192.168.1.193 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK77bb4337
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.1.219;tag=as5abee045
To: sip:192.168.1.193
Contact: sip:Unknown@192.168.1.219:5060
Call-ID: 5a47ba4e06ae70da5efb364d3e1cdbff@192.168.1.219:5060
CSeq: 102 OPTIONS
User-Agent: BrikerIPPBX
Date: Thu, 13 Apr 2017 08:25:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bK77bb4337
From: “Unknown” sip:Unknown@192.168.1.219;tag=as5abee045
To: sip:192.168.1.193;tag=6cf0c8f56d50f4e2
Call-ID: 5a47ba4e06ae70da5efb364d3e1cdbff@192.168.1.219:5060
CSeq: 102 OPTIONS
User-Agent: Grandstream GXW4108 (HW 1.1, Ch:8) 1.3.4.13
Contact: sip:5060@192.168.1.193:5060;transport=udp
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer, 100rel, path
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘5a47ba4e06ae70da5efb364d3e1cdbff@192.168.1.219:5060’ Method: OPTIONS
ippbx*CLI>

Mohon advicenya
Terima kasih

asoka · April 13, 2017, 8:30am

Coba lakukan yang ini, jangan dari asterisk console.

risa.futbas16 · April 13, 2017, 8:33am

Hallo,

saya sudah melakukan hal tersebut, dan log yang saya berikan adalah hasil dari perubahan setting tersebut.
terima kasih

asoka · April 13, 2017, 8:37am

Sebelum call lakukan perintah diatas, dan paste disini outputnya , jangan dari asterisk -rvvv

risa.futbas16 · April 13, 2017, 8:43am

hallo,

hasilnya hanya seperti ini :

terima kasih

asoka · April 13, 2017, 8:47am

IP Phone yang digunakan merk apa yah , sudah di cek konfigurasi dialplan pada IP Phonenya, kalo tidak terkirim ke asterisk berarti issuenya ada di IP Phone

risa.futbas16 · April 13, 2017, 8:57am

saya pakai ipphone fanvil C56
Untuk setting account sepert ini

dan setting dialplannya seperti ini

berarti ada issue pada ipphone nya ya?
terima kasih