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.