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標題: OBi110 being outbound trunks (GV, landline, OBiTalk) of Asterisk with one-stage dialling [打印本頁]

作者: 角色    時間: 2011-3-27 02:27     標題: OBi110 being outbound trunks (GV, landline, OBiTalk) of Asterisk with one-stage dialling

本帖最後由 角色 於 2011-3-27 02:43 編輯

In an Asterisk server, an extension 1910 is created as follows:

sip.conf
;TW OBi110 SP2
[1910]
type=friend
username=TW OBi110
secret=password
host=dynamic
canreinvite=no
dtmfmode = auto
disallow=all
allow=ulaw,alaw,gsm
context=internal

The dialling plans in Asterisk to access the voice resources in the OBi110 are shown below. Other extensions are not shown for the sake of simplicity.

extensions.conf
;111 OBi110 GV
[trunk-obi110-gv]
exten => _111.,1,Dial(SIP/**1${EXTEN:3}@1910,,)
exten => _111.,n,Hangup()

;118 OBi110 PSTN
[trunk-obi110-pstn]
exten => _118.,1,Dial(SIP/**8${EXTEN:3}@1910,,)
exten => _118.,n,Hangup()

;119 OBi110 OBiTalk
[trunk-obi110-obitalk]
exten => _119.,1,Dial(SIP/**9${EXTEN:3}@1910,,)
exten => _119.,n,Hangup()

[internal]
include => trunk-obi110-gv
include => trunk-obi110-pstn
include => trunk-obi110-obitalk

exten => 1910,1,Dial(SIP/1910,,) ;OBi110 SP2


In OBi110 Settings: (Use OBi Expert to edit the following variable)
Voice Service >> SP2 Service >> X_InboundCallRoute =
{@>(<**1:>xx.):sp1},{@>(<**8:>xx.):li},{@>(<**9:>xx.):pp}

Using one of extensions in Asterisk to carry out the following tests:

1) Key sequence to make US/CAN telephone call via the OBi110's GV gateway
Press 111 + <US/CAN telephone number>

2) Key sequence to make local PSTN calls via the OBi110's LINE port gateway
Press 118 + <PSTN call>

3) Key sequence to make OBiTalk calls via the OBi110's OBiTalk gateway
Press 119 + <PSTN call>


YH
作者: ckleea    時間: 2011-3-27 06:33

Thank, these are complete settings to interconnect OBi and asterisk
作者: bubblestar    時間: 2011-3-27 09:56

回復 1# 角色


   
OH! YH, you are really something.  Another great, clean and clear tutorial .  I won't miss this chance to merge it into my system.  Many thanks.
作者: ckleea    時間: 2011-3-27 15:57

回復 1# 角色


    I overlook your trunk setting and will try again later.
作者: ckleea    時間: 2011-3-27 17:33

An update, I am unable to make it work with asterisk 1.8
作者: 角色    時間: 2011-3-27 17:44

Have you updated the DigitMap for the SP2?

YH
作者: ckleea    時間: 2011-3-27 17:48

What is this DigitMap?
I follow your instruction on the first thread. On debug, I got 404 error.
This is the same result when I follow other thread in Internet

My other method works but 2 stage dialing.
作者: 角色    時間: 2011-3-27 18:16

Have you added the followings:

In OBi110 Settings: (Use OBi Expert to edit the following variable)
Voice Service >> SP2 Service >> X_InboundCallRoute =
{@>(<**1:>xx.):sp1},{@>(<**8:>xx.):li},{@>(<**9:>xx.):pp}

YH
作者: bubblestar    時間: 2011-3-27 18:27

I use YH's settings in my Asterisk.  All works as one-stage dialing except the **9 portion, which has to be routed via AA for further dialing (2-stage).  It is because that my Mobile phone number is included in my Trusted callers list to access AA.
作者: ckleea    時間: 2011-3-27 19:55

回復 8# 角色


    What is your setup for SP2? Just server name, user extensions and password?
作者: 角色    時間: 2011-3-27 20:11

回復 10# ckleea

Yes.
作者: ckleea    時間: 2011-3-27 21:26

I have made some changes to what you wrote
  1. [obitrunk]
  2. username = 6xxx
  3. type = peer
  4. secret = password
  5. host = 192.168.xxx.xxx
  6. fromdomain = 192.168.xxx.xxx
  7. fromuser = 6xxx
  8. context = from-obitrunk
  9. disallow = all
  10. allow = ulaw
  11. allow = alaw
  12. allow = gsm
  13. canreinvite = no
  14. port = 5061
  15. quality = yes
  16. nat = yes
  17. insecure = port,invite
複製代碼

作者: bubblestar    時間: 2011-3-27 21:29

回復 12# ckleea


    "quality" should read "qualify"
作者: ckleea    時間: 2011-3-27 21:39

Can you do speed dial?
作者: bubblestar    時間: 2011-3-27 21:58

The original speed dial prefix **8 cannot be used any longer as it is merged into the dialing rule in Asterisk extension.  What we will use now is our pre-defined prefix, such as 118X.  The advantage is that you can dial to whatever numbers you want with this rule in a flexible way.

I haven't tried the use of speed dial but I don't see why we can't.
作者: ckleea    時間: 2011-3-28 09:46

Speed dial still doesn't work.

Should it be **93?




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