Difference between revisions of "OpenBTS: Yate Config Test"

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Line 30: Line 30:
 
  [default]
 
  [default]
  
 +
; untuk call ke cell & pstn
 +
^08.*$=sip/sip:\0@192.168.0.4:5060
 +
^021.*$=sip/sip:\0@192.168.0.4:5060
 +
 +
atau
 +
 +
^08.*$=sip/sip:\0@192.168.0.5:5061
 +
^021.*$=sip/sip:\0@192.168.0.5:5061
  
; untuk call ke cell & pstn
 
^08\(.*\)$=sip/sip:2031@192.168.0.4/08\1
 
^021\(.*\)$=sip/sip:2031@192.168.0.4/021\1
 
  
  ; terakhir, semua telepon di approve
+
  ; terakhir, semua telepon di approve supaya tidak Unauthorized
 
  .*=return true
 
  .*=return true
  ${address}^192\.168\.0\.=return
+
 
 +
 
 +
===Beberapa Teknik Routing Outgoing call yang GAGAL TOTAL===
 +
 
 +
Teknik ini Gagal & jangan digunakan
 +
 
 +
  ^08\(.*\)$=sip/sip:2032@192.168.0.5:5061/08\1;tonedetect_out=yes
 +
^08.*$=sip/sip:2032@192.168.0.5:5061/\0
 +
 
 +
^021.*$=sip/sip:2032@192.168.0.5:5061/\0
 +
^021\(.*\)$=sip/sip:2032@192.168.0.5:5061/021\1;tonedetect_out=yes
 +
 
 +
tonedetect_out=yes tidak berefek apa-apa :( ..
 +
 
 +
 
 +
===Teknik Routing yang perlu di Test===
 +
 
 +
Kalau kita punya dua (2) Analog Telepon Adapter (ATA) ada baiknya coba fork
 +
 
 +
* Coba Dial Sekaligus
 +
 
 +
^08.*$=fork sip/sip:\0@192.168.0.5:5061 sip/sip:\0@192.168.0.4:5060;stoperror=busy
 +
^021.*$=fork sip/sip:\0@192.168.0.4:5060 sip/sip:\0@192.168.0.5:5061;stoperror=busy
 +
 
 +
 
 +
* Coba Dial Satu-Satu, kalau gagal semua kirim busy tone
 +
 
 +
^08.*$=fork sip/sip:\0@192.168.0.5:5061 | sip/sip:\0@192.168.0.4:5060;stoperror=busy
 +
^021.*$=fork sip/sip:\0@192.168.0.4:5060 | sip/sip:\0@192.168.0.5:5061;stoperror=busy
 +
 
 +
* Coba dial satu-satu, beri selang 3 detik
 +
 
 +
^08.*$=fork sip/sip:\0@192.168.0.5:5061 |next=3000 sip/sip:\0@192.168.0.4:5060;stoperror=busy
 +
^021.*$=fork sip/sip:\0@192.168.0.4:5060 |next=3000 sip/sip:\0@192.168.0.5:5061;stoperror=busy
 +
 
 +
==Referensi==
 +
 
 +
* http://yate.null.ro/pmwiki/index.php?n=Main.RoutingTips
 +
* http://yate.null.ro/pmwiki/index.php?n=Main.Routing
 +
* http://yate.null.ro/pmwiki/index.php?n=Main.Callfork
  
 
==Pranala Menarik==
 
==Pranala Menarik==

Revision as of 20:35, 5 August 2012

Konfigurasi untuk percobaan

regfile.conf

Edit

vi /etc/yate/regfile.conf 

Isi dengan

[2030]
password=123456
caller=2030

[2031]
password=123456
caller=2031

[2032]
password=123456
caller=2032


regexroute.conf

Edit

vi /etc/yate/regexroute.conf

Isi dengan

[default]
; untuk call ke cell & pstn
^08.*$=sip/sip:\0@192.168.0.4:5060
^021.*$=sip/sip:\0@192.168.0.4:5060

atau

^08.*$=sip/sip:\0@192.168.0.5:5061
^021.*$=sip/sip:\0@192.168.0.5:5061


; terakhir, semua telepon di approve supaya tidak Unauthorized
.*=return true


Beberapa Teknik Routing Outgoing call yang GAGAL TOTAL

Teknik ini Gagal & jangan digunakan

^08\(.*\)$=sip/sip:2032@192.168.0.5:5061/08\1;tonedetect_out=yes
^08.*$=sip/sip:2032@192.168.0.5:5061/\0
^021.*$=sip/sip:2032@192.168.0.5:5061/\0
^021\(.*\)$=sip/sip:2032@192.168.0.5:5061/021\1;tonedetect_out=yes

tonedetect_out=yes tidak berefek apa-apa :( ..


Teknik Routing yang perlu di Test

Kalau kita punya dua (2) Analog Telepon Adapter (ATA) ada baiknya coba fork

  • Coba Dial Sekaligus
^08.*$=fork sip/sip:\0@192.168.0.5:5061 sip/sip:\0@192.168.0.4:5060;stoperror=busy
^021.*$=fork sip/sip:\0@192.168.0.4:5060 sip/sip:\0@192.168.0.5:5061;stoperror=busy


  • Coba Dial Satu-Satu, kalau gagal semua kirim busy tone
^08.*$=fork sip/sip:\0@192.168.0.5:5061 | sip/sip:\0@192.168.0.4:5060;stoperror=busy
^021.*$=fork sip/sip:\0@192.168.0.4:5060 | sip/sip:\0@192.168.0.5:5061;stoperror=busy
  • Coba dial satu-satu, beri selang 3 detik
^08.*$=fork sip/sip:\0@192.168.0.5:5061 |next=3000 sip/sip:\0@192.168.0.4:5060;stoperror=busy
^021.*$=fork sip/sip:\0@192.168.0.4:5060 |next=3000 sip/sip:\0@192.168.0.5:5061;stoperror=busy

Referensi

Pranala Menarik

Persiapan

OpenBTS 2.6

OpenBTS 2.8

Multi OpenBTS 2.8

Ettus E110

Power Amplifier

Lain Lain

Catatan Legal dan Pendukung

Catatan Sejarah

Dokumentasi Video