Selamat siang Suhu,
Saya baru mencoba asterisk menggunakan FreePBX.
Problem saya adalah ketika melakukan panggilana maupun menerima lewat trunk Indihome kita tidak bisa mendengar suara dr remote, tp remote bisa mendengar suara kita.
Setelah saya coba cari informasi sebagian besar kemungkinan masalah NAT.
Tapi kalau masalah NAT ketika trunk Indihome saya pasang di Microsip di Laptop yg satu LAN dengan server FreePBX bisa lancar.
Antar extension baik dalam LAN maupun dr luar lewat port forwarding lancar juga.
Sudah berkutat selama 2 minggu tp belum berhasil juga.
Mohon bimbingannya.
Adapun topologinya:
Modem Indihome — Mikrotik — Server FreePBX
Setting di Microsip kalau coba trunk Indihome (hanya berfungsi ketika settingnya seperti di gambar, coba Zoiper di android belum bisa saja)
Configurasi Trunk di FreePBX:
username=+62xxxxxx@telkom.net.id
type=peer
supportpath=yes
secret=xxxxx
realm=telkom.net.id
qualify=yes
nat=force_rport,comedia
insecure=port,invite
host=10.0.0.10
fromuser=+62xxxxx
fromdomain=telkom.net.id
context=from-pstn
Configurasi sip FreePBX:
freepbx*CLI> sip show settings
Global Settings:
UDP Bindaddress: 0.0.0.0:5160
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
RTP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-14.0.2.14(15.2.2)
SDP Session Name: Asterisk PBX 15.2.2
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: NoNetwork QoS Settings:
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: Yes
Jitterbuffer forced: No
Jitterbuffer max size: 200
Jitterbuffer resync: 1000
Jitterbuffer impl: adaptive
Jitterbuffer tgt extra: 40
Jitterbuffer log: NoNetwork Settings:
SIP address remapping: Enabled using externhost
Externhost: xxx.xxxxxxx.com
Externaddr: 1xx.xx.xx.61:0
Externrefresh: 120
Localnet: 10.0.0.0/255.0.0.0
172.16.0.0/255.240.0.0
192.168.0.0/255.255.0.0
169.254.0.0/255.255.0.0Global Signalling Settings:
Codecs: (opus|alaw|ulaw)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 1
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 3600 secs
Reg. max duration: 3600 secs
Reg. default duration: 3600 secs
Sub. min duration 3600 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:No
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 1024 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70Default Settings:
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: en
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
RTCP Multiplexing: No
freepbx*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
104/104 192.168.10.2 D Yes Yes A 5060 OK (119 ms)
Indihome/+62xxxxxx@te 10.0.0.10 Yes Yes 5060 OK (68 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
freepbx*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
10.0.0.10:5060 N +62xxxxxx 3585 Registered Mon, 16 Apr 2018 13:13:17
1 SIP registrations.
Log ketika melakukan panggilan:
== Setting global variable ‘SIPDOMAIN’ to ‘192.168.10.8’
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Executing [9147@from-internal:1] Macro(“PJSIP/101-00000010”, “user-callerid,LIMIT”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/101-00000010”, “TOUCH_MONITOR=1523860352.43”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/101-00000010”, “AMPUSER=101”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/101-00000010”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/101-00000010”, “1?Set(REALCALLERIDNUM=101)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/101-00000010”, “AMPUSER=101”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/101-00000010”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“PJSIP/101-00000010”, “AMPUSERCIDNAME=Lenovo”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/101-00000010”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/101-00000010”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/101-00000010”, “AMPUSERCID=101”) in new stack
– Executing [s@macro-user-callerid:11] Set(“PJSIP/101-00000010”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/101-00000010”, “CALLERID(all)=“Lenovo” <101>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/101-00000010”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/101-00000010”, “1?Set(GROUP(concurrency_limit)=101)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“PJSIP/101-00000010”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“PJSIP/101-00000010”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/101-00000010”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] GotoIf(“PJSIP/101-00000010”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“PJSIP/101-00000010”, “CALLERID(number)=101”) in new stack
– Executing [s@macro-user-callerid:38] Set(“PJSIP/101-00000010”, “CALLERID(name)=Lenovo”) in new stack
– Executing [s@macro-user-callerid:39] GotoIf(“PJSIP/101-00000010”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:40] Set(“PJSIP/101-00000010”, “CDR(cnam)=Lenovo”) in new stack
– Executing [s@macro-user-callerid:41] Set(“PJSIP/101-00000010”, “CDR(cnum)=101”) in new stack
– Executing [s@macro-user-callerid:42] Set(“PJSIP/101-00000010”, “CHANNEL(language)=en”) in new stack
– Executing [9147@from-internal:2] Set(“PJSIP/101-00000010”, “ROUTEUSER=101”) in new stack
– Executing [9147@from-internal:3] Set(“PJSIP/101-00000010”, “ROUTEUSER=101”) in new stack
– Executing [9147@from-internal:4] GotoIf(“PJSIP/101-00000010”, “1?notblind”) in new stack
– Goto (from-internal,9147,7)
– Executing [9147@from-internal:7] GotoIf(“PJSIP/101-00000010”, “1?restrictedroute-c81e728d9d4c2f636f067f89cc14862c,9147,2:outbound-allroutes,9147,2”) in new stack
– Goto (restrictedroute-c81e728d9d4c2f636f067f89cc14862c,9147,2)
– Executing [9147@restrictedroute-c81e728d9d4c2f636f067f89cc14862c:2] Gosub(“PJSIP/101-00000010”, “sub-record-check,s,1(out,9147,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/101-00000010”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/101-00000010”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/101-00000010”, “NOW=1523860352”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/101-00000010”, “__DAY=16”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/101-00000010”, “__MONTH=04”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/101-00000010”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/101-00000010”, “__TIMESTR=20180416-133232”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/101-00000010”, “__FROMEXTEN=101”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/101-00000010”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/101-00000010”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/101-00000010”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/101-00000010”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/101-00000010”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/101-00000010”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/101-00000010”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“PJSIP/101-00000010”, “Outbound Recording Check from 101 to 9147”) in new stack
– Executing [out@sub-record-check:2] Set(“PJSIP/101-00000010”, “RECMODE=never”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“PJSIP/101-00000010”, “0?Goto(routewins)”) in new stack
– Executing [out@sub-record-check:4] ExecIf(“PJSIP/101-00000010”, “0?Goto(routewins)”) in new stack
– Executing [out@sub-record-check:5] Gosub(“PJSIP/101-00000010”, “recordcheck,1(never,out,9147)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/101-00000010”, “Starting recording check against never”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/101-00000010”, “never”) in new stack
– Goto (sub-record-check,recordcheck,14)
– Executing [recordcheck@sub-record-check:14] Set(“PJSIP/101-00000010”, “__REC_POLICY_MODE=NEVER”) in new stack
– Executing [recordcheck@sub-record-check:15] Goto(“PJSIP/101-00000010”, “stoprec”) in new stack
– Goto (sub-record-check,recordcheck,25)
– Executing [recordcheck@sub-record-check:25] NoOp(“PJSIP/101-00000010”, “Stopping recording: out, 9147”) in new stack
– Executing [recordcheck@sub-record-check:26] Set(“PJSIP/101-00000010”, “__REC_STATUS=STOPPED”) in new stack
– Executing [recordcheck@sub-record-check:27] System(“PJSIP/101-00000010”, "/var/lib/asterisk/bin/stoprecording.php “PJSIP/101-00000010"”) in new stack
– Executing [recordcheck@sub-record-check:28] Return(“PJSIP/101-00000010”, “”) in new stack
– Executing [out@sub-record-check:6] Return(“PJSIP/101-00000010”, “”) in new stack
– Executing [9147@restrictedroute-c81e728d9d4c2f636f067f89cc14862c:3] ExecIf(“PJSIP/101-00000010”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [9147@restrictedroute-c81e728d9d4c2f636f067f89cc14862c:4] Set(“PJSIP/101-00000010”, “MOHCLASS=default”) in new stack
– Executing [9147@restrictedroute-c81e728d9d4c2f636f067f89cc14862c:5] ExecIf(“PJSIP/101-00000010”, “1?Set(TRUNKCIDOVERRIDE=+62xxxxxxx2)”) in new stack
– Executing [9147@restrictedroute-c81e728d9d4c2f636f067f89cc14862c:6] Set(“PJSIP/101-00000010”, “_NODEST=”) in new stack
– Executing [9147@restrictedroute-c81e728d9d4c2f636f067f89cc14862c:7] Macro(“PJSIP/101-00000010”, “dialout-trunk,1,147,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“PJSIP/101-00000010”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“PJSIP/101-00000010”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:3] ExecIf(“PJSIP/101-00000010”, “0?Set(CALLERID(num)=101)”) in new stack
– Executing [s@macro-dialout-trunk:4] GotoIf(“PJSIP/101-00000010”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“PJSIP/101-00000010”, “DIAL_NUMBER=147”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“PJSIP/101-00000010”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“PJSIP/101-00000010”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“PJSIP/101-00000010”, “DIAL_TRUNK_OPTIONS=T”) in new stack
– Executing [s@macro-dialout-trunk:9] GotoIf(“PJSIP/101-00000010”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:10] GotoIf(“PJSIP/101-00000010”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:11] GotoIf(“PJSIP/101-00000010”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:12] Macro(“PJSIP/101-00000010”, “outbound-callerid,1”) in new stack
– Executing [s@macro-outbound-callerid:1] NoOp(“PJSIP/101-00000010”, “101”) in new stack
– Executing [s@macro-outbound-callerid:2] NoOp(“PJSIP/101-00000010”, “”) in new stack
– Executing [s@macro-outbound-callerid:3] NoOp(“PJSIP/101-00000010”, “all”) in new stack
– Executing [s@macro-outbound-callerid:4] ExecIf(“PJSIP/101-00000010”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:5] ExecIf(“PJSIP/101-00000010”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:6] ExecIf(“PJSIP/101-00000010”, “0?Set(REALCALLERIDNUM=101)”) in new stack
– Executing [s@macro-outbound-callerid:7] GotoIf(“PJSIP/101-00000010”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing [s@macro-outbound-callerid:11] Set(“PJSIP/101-00000010”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:12] Set(“PJSIP/101-00000010”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:13] Set(“PJSIP/101-00000010”, “TRUNKOUTCID=+62xxxxxxxx2”) in new stack
– Executing [s@macro-outbound-callerid:14] GotoIf(“PJSIP/101-00000010”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,19)
– Executing [s@macro-outbound-callerid:19] ExecIf(“PJSIP/101-00000010”, “1?Set(CALLERID(all)=+62xxxxxxx2)”) in new stack
– Executing [s@macro-outbound-callerid:20] ExecIf(“PJSIP/101-00000010”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:21] ExecIf(“PJSIP/101-00000010”, “1?Set(CALLERID(all)=+62xxxxxxx2)”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“PJSIP/101-00000010”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“PJSIP/101-00000010”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:24] Set(“PJSIP/101-00000010”, “CDR(outbound_cnum)=+62xxxxxxx2”) in new stack
– Executing [s@macro-outbound-callerid:25] Set(“PJSIP/101-00000010”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:13] GosubIf(“PJSIP/101-00000010”, “0?sub-flp-1,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“PJSIP/101-00000010”, “OUTNUM=147”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“PJSIP/101-00000010”, “custom=SIP/Indihome”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“PJSIP/101-00000010”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
– Executing [s@macro-dialout-trunk:17] ExecIf(“PJSIP/101-00000010”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:18] Macro(“PJSIP/101-00000010”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/101-00000010”, “”) in new stack
– Executing [s@macro-dialout-trunk:19] GotoIf(“PJSIP/101-00000010”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:20] Set(“PJSIP/101-00000010”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:21] Set(“PJSIP/101-00000010”, “__CRM_DESTINATION=147”) in new stack
– Executing [s@macro-dialout-trunk:22] Set(“PJSIP/101-00000010”, “__CRM_SOURCE=101”) in new stack
– Executing [s@macro-dialout-trunk:23] AGI(“PJSIP/101-00000010”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/101-00000010>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:24] Set(“PJSIP/101-00000010”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:25] NoOp(“PJSIP/101-00000010”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:26] GotoIf(“PJSIP/101-00000010”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/101-00000010”, “1?Set(CONNECTEDLINE(num,i)=147)”) in new stack
– Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/101-00000010”, “1?Set(CONNECTEDLINE(name,i)=CID:+62xxxxxxxx2)”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/101-00000010”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)+62xxxxxxxx2)”) in new stack
– Executing [s@macro-dialout-trunk:30] GotoIf(“PJSIP/101-00000010”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:31] Dial(“PJSIP/101-00000010”, “SIP/Indihome/147,300,Tb(func-apply-sipheaders^s^1)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– SIP/Indihome-0000001b Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/Indihome-0000001b”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] NoOp(“SIP/Indihome-0000001b”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“SIP/Indihome-0000001b”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“SIP/Indihome-0000001b”, “0”) in new stack
– Jumping to priority 8
– Executing [s@func-apply-sipheaders:9] Return(“SIP/Indihome-0000001b”, “”) in new stack
== Spawn extension (from-pstn, 9147, 1) exited non-zero on ‘SIP/Indihome-0000001b’
– SIP/Indihome-0000001b Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
Audio is at 17640
Adding codec opus to SDP
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.0.10:5060:
INVITE sip:147@10.0.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.8:5160;branch=z9hG4bK3c03e14c;rport
Max-Forwards: 70
From: sip:+62xxxxxxx2@telkom.net.id:5160;tag=as7ff984b3
To: sip:147@10.0.0.10
Contact: sip:+62xxxxxxx2@192.168.10.8:5160
Call-ID: 0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id
CSeq: 102 INVITE
User-Agent: FPBX-14.0.2.14(15.2.2)
Date: Mon, 16 Apr 2018 06:32:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer, path
Content-Type: application/sdp
Content-Length: 339v=0
o=root 1024073464 1024073464 IN IP4 192.168.10.8
s=Asterisk PBX 15.2.2
c=IN IP4 192.168.10.8
t=0 0
m=audio 17640 RTP/AVP 107 8 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 maxplaybackrate=24000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
-- Called SIP/Indihome/147<— SIP read from UDP:10.0.0.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.8:5160;branch=z9hG4bK3c03e14c;received=10.41.35.6;rport=5160
Call-ID: 0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id
From: sip:+62xxxxxxxx2@telkom.net.id:5160;tag=as7ff984b3
To: sip:147@10.0.0.10
CSeq: 102 INVITE
Content-Length: 0<------------->
— (7 headers 0 lines) —<— SIP read from UDP:10.0.0.10:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.8:5160;branch=z9hG4bK3c03e14c;received=10.41.35.6;rport=5160
Call-ID: 0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id
From: sip:+62xxxxxxxx2@telkom.net.id:5160;tag=as7ff984b3
To: sip:147@10.0.0.10;tag=sbc0905af9qf7s7-CC-71
CSeq: 102 INVITE
Contact: sip:10.0.0.10:5060
P-Early-Media: sendonly
Content-Length: 197
Content-Type: application/sdpv=0
o=- 74009762 74009762 IN IP4 10.0.0.2
s=SBC call
c=IN IP4 10.0.0.2
t=0 0
m=audio 30216 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (10 headers 10 lines) —
sip_route_dump: route/path hop: sip:10.0.0.10:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (opus|alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f681c026490 – Strict RTP learning after remote address set to: 10.0.0.2:30216
Peer audio RTP is at port 10.0.0.2:30216
– SIP/Indihome-0000001b is ringing
– SIP/Indihome-0000001b is making progress passing it to PJSIP/101-00000010
> 0x7f6820086030 – Strict RTP learning after remote address set to: 192.168.10.33:4002
> 0x7f6820086030 – Strict RTP learning after remote address set to: 192.168.10.33:4002
> 0x7f6820086030 – Strict RTP switching to RTP target address 192.168.10.33:4002 as source<— SIP read from UDP:10.0.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.8:5160;branch=z9hG4bK3c03e14c;received=10.41.35.6;rport=5160
Call-ID: 0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id
From: sip:+62xxxxxxx2@telkom.net.id:5160;tag=as7ff984b3
To: sip:147@10.0.0.10;tag=sbc0905af9qf7s7-CC-71
CSeq: 102 INVITE
Allow: INVITE,ACK,BYE,CANCEL,INFO,PRACK,NOTIFY,REFER,SUBSCRIBE,OPTIONS,MESSAGE
Contact: sip:10.0.0.10:5060
Require: timer
Session-Expires: 1800;refresher=uas
Content-Length: 197
Content-Type: application/sdpv=0
o=- 74009762 74009763 IN IP4 10.0.0.2
s=SBC call
c=IN IP4 10.0.0.2
t=0 0
m=audio 30216 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (12 headers 10 lines) —
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (opus|alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f681c026490 – Strict RTP learning after remote address set to: 10.0.0.2:30216
Peer audio RTP is at port 10.0.0.2:30216
sip_route_dump: route/path hop: sip:10.0.0.10:5060
Transmitting (NAT) to 10.0.0.10:5060:
ACK sip:10.0.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.8:5160;branch=z9hG4bK3ec3009b;rport
Max-Forwards: 70
From: sip:+62xxxxxxx2@telkom.net.id:5160;tag=as7ff984b3
To: sip:147@10.0.0.10;tag=sbc0905af9qf7s7-CC-71
Contact: sip:+62xxxxxxx2@192.168.10.8:5160
Call-ID: 0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id
CSeq: 102 ACK
User-Agent: FPBX-14.0.2.14(15.2.2)
Content-Length: 0
-- SIP/Indihome-0000001b answered PJSIP/101-00000010 -- Channel SIP/Indihome-0000001b joined 'simple_bridge' basic-bridge <c67523ef-50aa-41dc-8c54-426a69e0f333> -- Channel PJSIP/101-00000010 joined 'simple_bridge' basic-bridge <c67523ef-50aa-41dc-8c54-426a69e0f333>[2018-04-16 13:32:34] WARNING[25395][C-00000016]: chan_sip.c:8056 sip_indicate: Don’t know how to indicate condition 36
Really destroying SIP dialog ‘asbc4923rb3af7a3ieeqd37s7esb88r334fq@19500.0.ATS.jk1m-ats01.telkom.net.id.71’ Method: NOTIFY
> 0x7f6820086030 – Strict RTP learning complete - Locking on source address 192.168.10.33:4002
– Channel PJSIP/101-00000010 left ‘simple_bridge’ basic-bridge
== Spawn extension (macro-dialout-trunk, s, 31) exited non-zero on ‘PJSIP/101-00000010’ in macro ‘dialout-trunk’
== Spawn extension (restrictedroute-c81e728d9d4c2f636f067f89cc14862c, 9147, 7) exited non-zero on ‘PJSIP/101-00000010’
– Executing [h@restrictedroute-c81e728d9d4c2f636f067f89cc14862c:1] Hangup(“PJSIP/101-00000010”, “”) in new stack
== Spawn extension (restrictedroute-c81e728d9d4c2f636f067f89cc14862c, h, 1) exited non-zero on ‘PJSIP/101-00000010’
– PJSIP/101-00000010 Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/101-00000010”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/101-00000010”, “HANGUP CAUSE: 16”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/101-00000010”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/101-00000010”, “MASTER CHANNEL: 1523860352.43 = 1523860352.43”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/101-00000010”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/101-00000010”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/101-00000010”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– Channel SIP/Indihome-0000001b left ‘simple_bridge’ basic-bridge
Scheduling destruction of SIP dialog ‘0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id’ in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 10.0.0.10:5060:
BYE sip:10.0.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.8:5160;branch=z9hG4bK14f13257;rport
Max-Forwards: 70
From: sip:+62xxxxxxxxx2@telkom.net.id:5160;tag=as7ff984b3
To: sip:147@10.0.0.10;tag=sbc0905af9qf7s7-CC-71
Call-ID: 0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id
CSeq: 103 BYE
User-Agent: FPBX-14.0.2.14(15.2.2)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<— SIP read from UDP:10.0.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.8:5160;branch=z9hG4bK14f13257;received=10.41.35.6;rport=5160
Call-ID: 0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id
From: sip:+62xxxxxxxx2@telkom.net.id:5160;tag=as7ff984b3
To: sip:147@10.0.0.10;tag=sbc0905af9qf7s7-CC-71
CSeq: 103 BYE
Content-Length: 0<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘0f845b0d587028da2fb5a2b86be93dd2@telkom.net.id’ Method: INVITE
– <PJSIP/101-00000010>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/101-00000010”, “”) in new stack
== Spawn extension (restrictedroute-c81e728d9d4c2f636f067f89cc14862c, h, 1) exited non-zero on ‘PJSIP/101-00000010’
– PJSIP/101-00000010 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=