[ASK] Linksys SPA400 Briker - Tidak Bisa Call Dari Ext PABX ke Softphone


(Dimas Wisnu) #1

Hallo semua, salam kenal sebelumnya.
Saya sedang membangun VOIP di kantor, dimana Topologinya adalah :

PABX Panasonic - ext301, ext302
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Linksys Spa400
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Briker
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SoftPhone<3001> , SoftPhone<3002>

Ket : Ext 301 dan 302 Terhubung ke port FXO di linksys spa400

Kondisi saat ini : Dari SoftPhone<3001> , SoftPhone<3002> menelpon ke Ext PABX berjalan dengan baik.
Namun ketika dicoba menelepon dari Ext PABX conth Ext 101 menelpon ke Ext301 tidak diteruskan ke SoftPhone<3001> padahal sudah disetting Inbound nya.

Setting Trunk SPA400
PEER Detail

allow=ULAW&ALAW
call-limit=50
canreinvite=no
context=from-internal
nat=yes
disallow=all
host=192.168.8.53
insecure=very
secret=
type=peer
username=9000

Inbound Route Briker

DID kosong
Set Destination ke SoftPhone 3001

Mohon Pencerahanya om om semua, apa ada yang terlewat untuk setting inbound route nya, terima kasih


(godril) #2

ada baiknya jika disertakan log hasil dari test call tersebut. Biasanya ada di /var/log/asterisk/full atau /var/log/asterisk/messages .


(Dimas Wisnu) #3

Hallo Om,

saya coba dial dari extPabx ke softphone ada suara " The Number You’ve Dial Is Not In services "

saya tail -f /var/log/asterisk/messages keluar begini om

[Jan 8 19:54:14] NOTICE[1416] chan_sip.c: Registration from ‘941 sip:941@192.168.8.52:5060’ failed for ‘192.168.8.1:36560’ - Wrong password
[Jan 8 19:54:22] NOTICE[1416] chan_sip.c: Registration from ‘951 sip:951@192.168.8.52:5060’ failed for ‘192.168.8.1:34486’ - Wrong password
[Jan 8 19:54:29] NOTICE[1416] chan_sip.c: Registration from ‘961 sip:961@192.168.8.52:5060’ failed for ‘192.168.8.1:5066’ - Wrong password
[Jan 8 19:54:37] NOTICE[1416] chan_sip.c: Registration from ‘971 sip:971@192.168.8.52:5060’ failed for ‘192.168.8.1:50616’ - Wrong password
[Jan 8 19:54:44] NOTICE[1416] chan_sip.c: Registration from ‘981 sip:981@192.168.8.52:5060’ failed for ‘192.168.8.1:5092’ - Wrong password
[Jan 8 19:54:51] NOTICE[1416] chan_sip.c: Registration from ‘466 sip:466@192.168.8.52:5060’ failed for ‘192.168.8.1:59064’ - Wrong password
[Jan 8 19:54:58] NOTICE[1416] chan_sip.c: Registration from ‘1115 sip:1115@192.168.8.52:5060’ failed for ‘192.168.8.1:5072’ - Wrong password
[Jan 8 19:55:05] NOTICE[1416] chan_sip.c: Registration from ‘1110 sip:1110@192.168.8.52:5060’ failed for ‘192.168.8.1:43711’ - Wrong password
[Jan 8 19:55:14] NOTICE[1416] chan_sip.c: Registration from ‘1212 sip:1212@192.168.8.52:5060’ failed for ‘192.168.8.1:1105’ - Wrong password
[Jan 8 19:55:20] NOTICE[1416] chan_sip.c: Registration from ‘1313 sip:1313@192.168.8.52:5060’ failed for ‘192.168.8.1:1144’ - Wrong password
[Jan 8 19:55:28] NOTICE[1416] chan_sip.c: Registration from ‘1414 sip:1414@192.168.8.52:5060’ failed for ‘192.168.8.1:40321’ - Wrong password
[Jan 8 19:55:34] NOTICE[1416] chan_sip.c: Registration from ‘1515 sip:1515@192.168.8.52:5060’ failed for ‘192.168.8.1:5066’ - Wrong password
[Jan 8 19:55:41] NOTICE[1416] chan_sip.c: Registration from ‘1616 sip:1616@192.168.8.52:5060’ failed for ‘192.168.8.1:39700’ - Wrong password
[Jan 8 19:55:49] NOTICE[1416] chan_sip.c: Registration from ‘1717 sip:1717@192.168.8.52:5060’ failed for ‘192.168.8.1:1116’ - Wrong password
[Jan 8 19:55:56] NOTICE[1416] chan_sip.c: Registration from ‘1818 sip:1818@192.168.8.52:5060’ failed for ‘192.168.8.1:1114’ - Wrong password
[Jan 8 19:56:04] NOTICE[1416] chan_sip.c: Registration from ‘1919 sip:1919@192.168.8.52:5060’ failed for ‘192.168.8.1:59576’ - Wrong password
[Jan 8 19:56:11] NOTICE[1416] chan_sip.c: Registration from ‘2121 sip:2121@192.168.8.52:5060’ failed for ‘192.168.8.1:1112’ - Wrong password
[Jan 8 19:56:18] NOTICE[1416] chan_sip.c: Registration from ‘2323 sip:2323@192.168.8.52:5060’ failed for ‘192.168.8.1:5091’ - Wrong password


(Dimas Wisnu) #4
kalau pake CLI nya Asterisk -rvvvvvv keluar begini om telpon dari ExtPabx ke Softphone
> <--- SIP read from UDP:192.168.8.53:5060 --->
NOTIFY sip:9000@:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK38f947cb
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>
Contact: <sip:SPA400@192.168.8.53>
Call-ID: 474d9ecd0717461509f5f50b738fdb63@192.168.8.53
CSeq: 619 NOTIFY
User-Agent: LINKSYS/SPA400
Event: fxo-port-state;partial;1=inuse,v=28
Content-Length: 0



<------------->
--- (10 headers 1 lines) ---
Sending to 192.168.8.53:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.8.53:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK38f947cb;received=192.168.8.53
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>;tag=as190eb612
Call-ID: 474d9ecd0717461509f5f50b738fdb63@192.168.8.53
CSeq: 619 NOTIFY
Server: BrikerIPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '474d9ecd0717461509f5f50b738fdb63@192.168.8.53' in 32000 ms (Method: NOTIFY)

<--- SIP read from UDP:192.168.8.53:5060 --->
NOTIFY sip:9000@:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK7d43467f
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>
Contact: <sip:SPA400@192.168.8.53>
Call-ID: 56d5a637012ecc3f3a4ce025710d1a07@192.168.8.53
CSeq: 620 NOTIFY
User-Agent: LINKSYS/SPA400
Event: fxo-port-state;partial;1=ring,v=28
Content-Length: 0



<------------->
--- (10 headers 1 lines) ---
Sending to 192.168.8.53:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.8.53:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK7d43467f;received=192.168.8.53
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>;tag=as1a275ec7
Call-ID: 56d5a637012ecc3f3a4ce025710d1a07@192.168.8.53
CSeq: 620 NOTIFY
Server: BrikerIPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '56d5a637012ecc3f3a4ce025710d1a07@192.168.8.53' in 32000 ms (Method: NOTIFY)
    -- Registered SIP '3001' at 192.168.8.220:5060
    -- Registered SIP '3001' at 192.168.8.1:1149
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1

<--- SIP read from UDP:192.168.8.53:5060 --->
INVITE sip:9000@192.168.8.52 SIP/2.0
From: - 3311<sip:anonymous@localhost>;tag=3508a8c0-13c4-3d7ec84e-ddc9b1f-78bccb7
To: <sip:3311@192.168.8.53>
Call-ID: 10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK-3d7ec84e-ddc9b1f-1b701c24
Max-Forwards: 70
Supported: replaces,timer,100rel
Contact: <sip:3311@192.168.8.53:5060;transport=UDP>
Content-Type: application/SDP
Content-Length: 288

v=0
o=anonymous 8884 6672 IN IP4 192.168.8.53
s=SIP Call
c=IN IP4 192.168.8.53
t=0 0
m=audio 10002 RTP/AVP 8 101 0 18
a=ptime:30
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=sendrecv
<------------->
--- (11 headers 14 lines) ---
Sending to 192.168.8.53:5060 (no NAT)
Sending to 192.168.8.53:5060 (no NAT)
Using INVITE request as basis request - 10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost
Found peer 'spa400' for 'anonymous' from 192.168.8.53:5060
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 0
Found RTP audio format 18
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Capabilities: us - (alaw|ulaw|vp8|h264|h263p|h263), peer - audio=(ulaw|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 192.168.8.53:10002
Peer doesn't provide video
Peer doesn't provide T.140
Looking for 9000 in from-sip-external (domain 192.168.8.52)
sip_route_dump: route/path hop: <sip:3311@192.168.8.53:5060;transport=UDP>

<--- Transmitting (no NAT) to 192.168.8.53:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK-3d7ec84e-ddc9b1f-1b701c24;received=192.168.8.53
From: - 3311<sip:anonymous@localhost>;tag=3508a8c0-13c4-3d7ec84e-ddc9b1f-78bccb7
To: <sip:3311@192.168.8.53>
Call-ID: 10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost
CSeq: 1 INVITE
Server: BrikerIPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:9000@192.168.8.52:5060>
Content-Length: 0


<------------>
    -- Executing [9000@from-sip-external:1] NoOp("SIP/spa400-00002984", "Received incoming SIP connection from unknown peer to 9000") in new stack
    -- Executing [9000@from-sip-external:2] Set("SIP/spa400-00002984", "DID=9000") in new stack
    -- Executing [9000@from-sip-external:3] Goto("SIP/spa400-00002984", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/spa400-00002984", "0?from-trunk,9000,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/spa400-00002984", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2018-01-08 19:57:56.889 WIB.
    -- Executing [s@from-sip-external:3] Answer("SIP/spa400-00002984", "") in new stack
Audio is at 12948
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.8.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK-3d7ec84e-ddc9b1f-1b701c24;received=192.168.8.53
From: - 3311<sip:anonymous@localhost>;tag=3508a8c0-13c4-3d7ec84e-ddc9b1f-78bccb7
To: <sip:3311@192.168.8.53>;tag=as2fb81554
Call-ID: 10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost
CSeq: 1 INVITE
Server: BrikerIPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:9000@192.168.8.52:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 263

v=0
o=root 1554670896 1554670896 IN IP4 192.168.8.52
s=Asterisk PBX 13.9.1
c=IN IP4 192.168.8.52
t=0 0
m=audio 12948 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

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

<--- SIP read from UDP:192.168.8.53:5060 --->
ACK sip:9000@192.168.8.52:5060 SIP/2.0
From: <sip:anonymous@localhost>;tag=3508a8c0-13c4-3d7ec84e-ddc9b1f-78bccb7
To: <sip:3311@192.168.8.53>;tag=as2fb81554
Call-ID: 10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK-3d7ec84e-ddc9b97-3f27238d
Max-Forwards: 70
Contact: <sip:3311@192.168.8.53:5060;transport=UDP>
Content-Length: 0

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

<--- SIP read from UDP:192.168.8.53:5060 --->
NOTIFY sip:9000@:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK3013a7f3
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>
Contact: <sip:SPA400@192.168.8.53>
Call-ID: 283faa4d68233d434ac8b3f12d6c8d80@192.168.8.53
CSeq: 621 NOTIFY
User-Agent: LINKSYS/SPA400
Event: fxo-port-state;partial;1=off,v=28
Content-Length: 0



<------------->
--- (10 headers 1 lines) ---
Sending to 192.168.8.53:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.8.53:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK3013a7f3;received=192.168.8.53
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>;tag=as5a768234
Call-ID: 283faa4d68233d434ac8b3f12d6c8d80@192.168.8.53
CSeq: 621 NOTIFY
Server: BrikerIPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '283faa4d68233d434ac8b3f12d6c8d80@192.168.8.53' in 32000 ms (Method: NOTIFY)
       > 0x7f9109523840 -- Probation passed - setting RTP source address to 192.168.8.53:10002
    -- Executing [s@from-sip-external:4] Wait("SIP/spa400-00002984", "2") in new stack

<--- SIP read from UDP:192.168.8.53:5060 --->
NOTIFY sip:9000@:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK06af1767
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>
Contact: <sip:SPA400@192.168.8.53>
Call-ID: 5a0c8ee61644e7a84583609e575cb903@192.168.8.53
CSeq: 622 NOTIFY
User-Agent: LINKSYS/SPA400
Event: fxo-port-state;partial;1=off,v=7
Content-Length: 0



<------------->
--- (10 headers 1 lines) ---
Sending to 192.168.8.53:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.8.53:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK06af1767;received=192.168.8.53
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>;tag=as1065636e
Call-ID: 5a0c8ee61644e7a84583609e575cb903@192.168.8.53
CSeq: 622 NOTIFY
Server: BrikerIPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5a0c8ee61644e7a84583609e575cb903@192.168.8.53' in 32000 ms (Method: NOTIFY)
    -- Executing [s@from-sip-external:5] Playback("SIP/spa400-00002984", "ss-noservice") in new stack
    -- <SIP/spa400-00002984> Playing 'ss-noservice.slin' (language 'en')
  == Manager 'admin' logged on from 127.0.0.1
    -- Executing [s@from-sip-external:6] PlayTones("SIP/spa400-00002984", "congestion") in new stack
    -- Executing [s@from-sip-external:7] Congestion("SIP/spa400-00002984", "5") in new stack
  == Manager 'admin' logged off from 127.0.0.1
  == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/spa400-00002984'
    -- Executing [h@from-sip-external:1] NoOp("SIP/spa400-00002984", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/spa400-00002984", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/spa400-00002984", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/spa400-00002984", "0?from-trunk,s,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/spa400-00002984", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2018-01-08 19:58:08.992 WIB.
    -- Executing [s@from-sip-external:3] Answer("SIP/spa400-00002984", "") in new stack
  == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/spa400-00002984'
Scheduling destruction of SIP dialog '10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:3311@192.168.8.53:5060;transport=UDP> for address/port to send to
set_destination: set destination to 192.168.8.53:5060
Reliably Transmitting (no NAT) to 192.168.8.53:5060:
BYE sip:3311@192.168.8.53:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.52:5060;branch=z9hG4bK6eeaf0c8
Max-Forwards: 70
From: <sip:3311@192.168.8.53>;tag=as2fb81554
To: - 3311<sip:anonymous@localhost>;tag=3508a8c0-13c4-3d7ec84e-ddc9b1f-78bccb7
Call-ID: 10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost
CSeq: 102 BYE
User-Agent: BrikerIPPBX
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---

<--- SIP read from UDP:192.168.8.53:5060 --->
SIP/2.0 200 OK
From: <sip:3311@192.168.8.53>;tag=as2fb81554
To: <sip:anonymous@localhost>;tag=3508a8c0-13c4-3d7ec84e-ddc9b1f-78bccb7
Call-ID: 10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost
CSeq: 102 BYE
Via: SIP/2.0/UDP 192.168.8.52:5060;branch=z9hG4bK6eeaf0c8
Supported: replaces,timer,100rel
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '10195414-3508a8c0-13c4-3d7ec84e-ddc9b1f-710ca351@localhost' Method: ACK

<--- SIP read from UDP:192.168.8.53:5060 --->
NOTIFY sip:9000@:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK3d0ece53
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>
Contact: <sip:SPA400@192.168.8.53>
Call-ID: 2f0c2f412a5099394704c35e229c0aa5@192.168.8.53
CSeq: 623 NOTIFY
User-Agent: LINKSYS/SPA400
Event: fxo-port-state;partial;1=on,v=7
Content-Length: 0



<------------->
--- (10 headers 1 lines) ---
Sending to 192.168.8.53:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.8.53:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK3d0ece53;received=192.168.8.53
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>;tag=as399e2b5c
Call-ID: 2f0c2f412a5099394704c35e229c0aa5@192.168.8.53
CSeq: 623 NOTIFY
Server: BrikerIPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2f0c2f412a5099394704c35e229c0aa5@192.168.8.53' in 32000 ms (Method: NOTIFY)

<--- SIP read from UDP:192.168.8.53:5060 --->
NOTIFY sip:9000@:0 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK2793dfb8
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>
Contact: <sip:SPA400@192.168.8.53>
Call-ID: 1dda699623cad6e461e0bfdd0ee7839d@192.168.8.53
CSeq: 624 NOTIFY
User-Agent: LINKSYS/SPA400
Event: fxo-port-state;partial;1=on,v=28
Content-Length: 0



<------------->
--- (10 headers 1 lines) ---
Sending to 192.168.8.53:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.8.53:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.8.53:5060;branch=z9hG4bK2793dfb8;received=192.168.8.53
From: "SPA400" <sip:SPA400@192.168.8.53>;tag=as327b23c6
To: <sip:9000@:0>;tag=as76836b77
Call-ID: 1dda699623cad6e461e0bfdd0ee7839d@192.168.8.53
CSeq: 624 NOTIFY
Server: BrikerIPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

(Dimas Wisnu) #5

sudah done mas @godril untuk incoming dan outgoing
saya ubah trunk nya menjadi

host=192.168.8.53
username=9000
context=from-trunk
type=peer
qualify=yes
nat=no
directmedia=no
insecure=port,invite
disallow=all
allow=alaw&ulaw
transport=udp

yang jadi maalah sekarang gimana cara setting incoming untuk masing2 port fxo linksys diarahkan ke softphone yang berbeda beda, karena saya test call dari extpabx ke softphone yang keluar namanya anonymous dan no - 11301

fyi : port id linksys spa400 : port 1 = 11301, port 2 = 11302