Dial patterns di asterisk tidak berfungsi ketika di gateway

Asterisk
1

Halo mas mas sekalian member-member ID Asterisk, saya ingin menanyakan sesuatu yaitu dial patterns yang tidak berfungsi di asterisk ketika di gateway jadi problem ini cukup baru saya temui karna sebelum nya tidak pernah mendapatkan masalah di dalam setup ini, jadi begini mekanisme nya

VOS (Softswitch) -> Asterisk -> Gateway

Client dial 6200 terus di translate lewat VOS tersebut ke asterisk menjadi 520 jadi ketika ingin melakukan test dari sisi saya bukan dari client saya menggunakan 52062857xxxx misal nya harus nya bisa langsung call tetapi prefix atau dial patterns yang ada di asterisk tidak berfungsi, berikut saya pastekan log dari gateway nya serta screenshoot dial patterns nya.

INVITE sip:5200857xxxxxxxx@192.26.0.65 SIP/2.0
Via: SIP/2.0/UDP 192.26.0.109:5060;branch=z9hG4bK69f3627d;rport
Max-Forwards: 70
From: “1000” sip:1000@192.26.0.109;tag=as2fd98dbe
To: sip:5200857xxxxxxxx@192.26.0.65
Contact: sip:1000@192.26.0.109:5060
Call-ID: 3f83ca162b25f4f85745feae6ebb048c@192.26.0.109:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Tue, 13 Jun 2017 10:04:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370

v=0
o=root 1727241397 1727241397 IN IP4 192.26.0.109
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.26.0.109
t=0 0
m=audio 12134 RTP/AVP 0 8 3 9 112 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

01/01 00:56:21.293|6631| INFO| 0|sip_adaptor:3700| -| SIP| transaction 241 proccess event RCV_REQINVITE
01/01 00:56:21.293|6632| INFO| 0|sip_adaptor:3700| -| SIP|initialize session timer … (did = 169, expires = 0)
01/01 00:56:21.293|6633| INFO| 0|sip_adaptor:3700| -| SIP|Session-Expires header required(cid = 168, did = 169)
01/01 00:56:21.294|6634| INFO| 0|sip_adaptor:1253| -| SIP|url: “1000” sip:1000@192.26.0.109;tag=as2fd98dbe
01/01 00:56:21.295|6635| INFO| 0|sip_adaptor:1253| -| SIP|url: sip:5200857xxxxxxxx@192.26.0.65;soid=35;tag=40ebd492
01/01 00:56:21.295|6636| INFO| 0|sip_adaptor:1674| -| SIP|caller: “1000” 1000, called: “” 5200857xxxxxxxx
01/01 00:56:21.295|6637| INFO| 0|sip_adaptor:1759| -| SIP|sip route info: sock = 35, caller = 1000, callee = 5200857xxxxxxxx, server = 192.26.0.109:5060
01/01 00:56:21.298|6638| INFO| 0|sip_adaptor:1447| -| SIP|phone 1 send new call “3f83ca162b25f4f85745feae6ebb048c@192.26.0.109:5060”
01/01 00:56:21.298|6639| INFO| 0|sip_client.:0759| -| SIP|clear the endpoint 1 status 0X00003FE0!
01/01 00:56:21.298|6640| INFO| 0|sip_client.:0763| -| SIP|set the endpoint 1 status 0X00001000!
01/01 00:56:21.299|6641| INFO| 0|sip_client.:0885| -| SIP|send message SETUP to phone 1!
01/01 00:56:21.299|6642| INFO| 0|sip_client.:0690| -| SIP|state of endpoint 1: BUSY <- IDLE
01/01 00:56:21.302|6643| INFO| 0|sip_adaptor:3700| -| SIP| transaction 241 state IST_PROCEEDING
01/01 00:56:21.302|6644| INFO| 0|sip_adaptor:0954| -| SIP|PipeMsg: jpipe_call_answer(0x2acedd70, 183, 16870)!
01/01 00:56:21.304|6645| INFO| 0|sip_adaptor:3700| -| SIP| transaction 241 proccess event SND_STATUS_1XX
01/01 00:56:21.305|6646| INFO| 0|sip_adaptor:3642| -| SIP| Send to 192.26.0.109:5060 1970-01-01 00:56:21

Terima kasih banyak sebelum nya mohon dibantu.

2

Halo,

+62|. = berarti dialpatternnya harus match +62, setelah itu +62 nya akan dipotong dan dikirim yang setelahnya

begitupun 5200|. = itu berarti akan potong 5200, dan kirim sisanya

Jadi kalau mau dari User di Asterisk bisa langsung dial ke Gateway, cukup isi pada match pattern 520X.
kecuali 520 nya mau dipotong, maka 520 | X.

3

halo mas terimakasih sudah jawab,
jadi gini mas kalo saya manggil 520 di sip log nya 520 nya di translated sebagai nomor aneh nya, kalau seperti itu kira2 ada yang salah dimana nya ya mas

4

nomer aneh itu yang mana mas?

5

dial patterns nya yang digambar atas mas, jadi kalo kita call 52008578129312 misal nya harus nya jadi nya langsung 08578129312 seperti itu kan mas ? nah ini malah jadi 520 nya ikut gabung jadi deh reason 503 atau nomor salah

6

boleh ndak mas semua percakapan callid 3f83ca162b25f4f85745feae6ebb048c dipaste ke sini.

7

inimas

INVITE sip:520085710042008@172.26.0.65 SIP/2.0
Via: SIP/2.0/UDP 172.26.0.109:5060;branch=z9hG4bK75ad6b47;rport
Max-Forwards: 70
From: “1000” sip:1000@172.26.0.109;tag=as1881af7b
To: sip:520085710042008@172.26.0.65
Contact: sip:1000@172.26.0.109:5060
Call-ID: 6ed177db2e7f244e45b852b659bcda55@172.26.0.109:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 14 Jun 2017 04:40:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370

v=0
o=root 1182809311 1182809311 IN IP4 172.26.0.109
s=Asterisk PBX 1.8.20.0
c=IN IP4 172.26.0.109
t=0 0
m=audio 16304 RTP/AVP 0 8 3 9 112 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

01/01 19:32:10.423|58240| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 proccess event RCV_REQINVITE
01/01 19:32:10.423|58241| INFO| 0|sip_adaptor:3700| -| SIP|initialize session timer … (did = 2077, expires = 0)
01/01 19:32:10.423|58242| INFO| 0|sip_adaptor:3700| -| SIP|Session-Expires header required(cid = 2076, did = 2077)
01/01 19:32:10.424|58243| INFO| 0|sip_adaptor:1253| -| SIP|url: “1000” sip:1000@172.26.0.109;tag=as1881af7b
01/01 19:32:10.425|58244| INFO| 0|sip_adaptor:1253| -| SIP|url: sip:520085710042008@172.26.0.65;soid=35;tag=36e6c06c
01/01 19:32:10.425|58245| INFO| 0|sip_adaptor:1674| -| SIP|caller: “1000” 1000, called: “” 520085710042008
01/01 19:32:10.425|58246| INFO| 0|sip_adaptor:1759| -| SIP|sip route info: sock = 35, caller = 1000, callee = 520085710042008, server = 172.26.0.109:5060
01/01 19:32:10.428|58247|ERROR| 12|sip_adaptor:1785| -| SIP|no available pstn line! idles= 32
01/01 19:32:10.428|58248| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 state IST_PROCEEDING
01/01 19:32:10.429|58249| INFO| 0|sip_adaptor:0954| -| SIP|PipeMsg: jpipe_call_answer(0x2acedd70, 503, 0)!
01/01 19:32:10.429|58250| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 proccess event SND_STATUS_3456XX
01/01 19:32:10.432|58251| INFO| 0|sip_adaptor:3642| -| SIP| Send to 172.26.0.109:5060 1970-01-01 19:32:10

SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.26.0.109:5060;branch=z9hG4bK75ad6b47;rport=5060
From: “1000” sip:1000@172.26.0.109;tag=as1881af7b
To: sip:520085710042008@172.26.0.65;soid=35;tag=36e6c06c
Call-ID: 6ed177db2e7f244e45b852b659bcda55@172.26.0.109:5060
CSeq: 102 INVITE
User-Agent: EJOIN ACOM532 V4.7 r7824 b0510
Content-Length: 0

01/01 19:32:10.433|58252| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 state IST_COMPLETED
01/01 19:32:10.460|58253| INFO| 0|sip_adaptor:3678| -| SIP| Received from 172.26.0.109:5060 1970-01-01 19:32:10

ACK sip:520085710042008@172.26.0.65 SIP/2.0
Via: SIP/2.0/UDP 172.26.0.109:5060;branch=z9hG4bK75ad6b47;rport
Max-Forwards: 70
From: “1000” sip:1000@172.26.0.109;tag=as1881af7b
To: sip:520085710042008@172.26.0.65;tag=36e6c06c
Contact: sip:1000@172.26.0.109:5060
Call-ID: 6ed177db2e7f244e45b852b659bcda55@172.26.0.109:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0

01/01 19:32:10.462|58254| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 proccess event RCV_REQACK
01/01 19:32:10.462|58255| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 state IST_CONFIRMED
01/01 19:32:11.340|58256| INFO| 0|sip_adaptor:3642| -| SIP| Send to 172.26.0.109:5060 1970-01-01 19:32:11

01/01 19:32:12.220|58257| INFO| 0|ccm_wmctl.c:4754| -| CCM|port 1 card 0 lock base timeout!
01/01 19:32:12.220|58258| INFO| 0|ccm_wmctl.c:5268| -| CCM|WMCTL: send cmd UPDATE_BS_INFO! port = 1A, state = LOCK_BASE
01/01 19:32:12.221|58259| INFO| 0|ccm_wmctl.c:5750| -| CCM|WMCTL: disable card … port = 1, status = 2, reason = Update Base Info …
01/01 19:32:12.221|58260| INFO| 0|ccm_wmctl.c:5489| -| CCM|WMCTL: state changed! port = 1, state = 11 QUERY_BASE
01/01 19:32:12.480|58261| INFO| 0|ccm_wmctl.c:2610| -| CCM|WMCTL: got base station info! port = 1, bsi = 632;-84,0;0,0;0,0;0,0;0,0;0,0;0, state = QUERY_BASE, flags = 20
01/01 19:32:12.480|58262| INFO| 0|ccm_wmctl.c:5727| -| CCM|WMCTL: enable port … port = 1
01/01 19:32:12.481|58263| INFO| 0|ccm_wmctl.c:5783| -| CCM|WMCTL: enable port … port = 1
01/01 19:32:12.481|58264| INFO| 0|ccm_wmctl.c:5489| -| CCM|WMCTL: state changed! port = 1, state = 7 WORKING
01/01 19:32:12.482|58265| INFO| 0|ccm_wmctl.c:5290| -| CCM|port 1 slot 0 will update base station in 3600 seconds!
01/01 19:32:15.560|58266| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 proccess event TIMEOUT_I
01/01 19:32:15.560|58267| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 state IST_TERMINATED
01/01 19:32:15.561|58268| INFO| 0|sip_adaptor:3700| -| SIP|free session timer … (did = 2077, expires = 0)
01/01 19:32:15.561|58269| CRIT| -1|osip_transa:0290| -| SIP|transaction already removed from list 4288!
01/01 19:32:15.561|58270| INFO| 0|sip_adaptor:3700| -| SIP| transaction 4288 free ressource 6ed177db2e7f244e45b852b659bcda55
01/01 19:32:30.460|58271| INFO| 0|sip_adaptor:3642| -| SIP| Send to 172.26.0.109:5060 1970-01-01 19:32:30

8

Coba kirim lagi dial pattern yang terakhir, sudah mengikuti yang sarankan atau belum , yaitu 520 | X.

1 Like
9

jadi kalo saya restore ke konfigurasi sebelum nya bisa mas berfungsi si dialpatterns nya tapi besok hari nya sudah ga bisa lagi, balik lagi permasalahan nya :sweat_smile: