November 13, 2018, 12:38:19 pm *
Welcome, Guest. Please login or register.
News:
 
   Forum Home   Search Login Register OBiTALK  
Pages: [1]
  Print  
Author Topic: help i want to bridge sp1(Provider Guatemala number) to sp2 (extention)  (Read 2389 times)
Russ2709
Newbie
*
Posts: 2


« on: June 19, 2012, 01:08:34 am »

Thank you all for this wonderful forum!

I will explain my setup and what i would like to do (if possible):

- I am in Guatemala
- I have an Obi110, with sp1 whit a server from guatemala and
- the Obi110 also has a Sp2 it an extention of an server whit asterisk of my own

I would like to:

1. Call my local line (Guatemala line thats connected to the Obi110) from sp1 Number of guatemala and have the Obi110 answer after "X" amount of rings (lets say 10 rings) and have the Obi give me dial tone from sp2 (extention of my server)

2. When i use another extention to call the extention on my obi I like to be able to call from SP1 guatemala number from anoter extention of my own  whitout the auto atental just the sp1 give me dial tone too.

Im really sorry if i over explained it or was redundant, i just wanted to be as clear as possible to see if i can get a correct answer.

Maybe im expecting too much from the Obi110, but i think it could be possible.

Thank you very much in advance for your help.
Logged
Stewart
Hero Member
*****
Posts: 1125


« Reply #1 on: June 19, 2012, 03:27:03 am »

If you have a landline (PSTN) connected to the Line port of your OBi, you can use it as a trunk on Asterisk to make calls, without the auto-attendant.  See http://www.freepbx.org/support/documentation/howtos/howto-use-an-obi-110-device-to-provide-to-allow-freepbx-to-make-calls-o (ignore the part about Google Voice). 

Incoming calls could be sent to Asterisk's DISA if you like, but you would want to prompt for a password before giving dial tone; otherwise anyone calling your number could make expensive calls through your Asterisk.  If you want a delay before sending the call to Asterisk, add it to the InboundCallRoute for the Line port; see the section 1.3.0(2575) in http://www.obitalk.com/forum/index.php?topic=9.0

Sorry, but I don't completely understand your question.  What service do you currently have on SP1? If this is a Guatemala SIP service, why don't you use it with Asterisk directly?  Does it have a DID, separate from the number on your landline?

Are you using FreePBX or other GUI with Asterisk?  If so, which one?
Logged
Russ2709
Newbie
*
Posts: 2


« Reply #2 on: June 27, 2012, 09:42:57 pm »

Thank you all for this wonderful forum!

I will explain my setup and what i would like to do (if possible):

- I am in Guatemala
- I have an Obi110, with sp1 whit a server from guatemala and
- the Obi110 also has a Sp2 it an extention of an server whit asterisk of my own

I would like to:

1. Call my local line (Guatemala line thats connected to the Obi110) from sp1 Number of guatemala and have the Obi110 answer after "X" amount of rings (lets say 10 rings) and have the Obi give me dial tone from sp2 (extention of my server)

2. When i use another extention to call the extention on my obi I like to be able to call from SP1 guatemala number from anoter extention of my own  whitout the auto atental just the sp1 give me dial tone too.

Im really sorry if i over explained it or was redundant, i just wanted to be as clear as possible to see if i can get a correct answer.

Maybe im expecting too much from the Obi110, but i think it could be possible.

Thank you very much in advance for your help.

no no you dont understand i dont wish to use pstn just sp1 and sp2 uso a thrirth phone to call sp1 then sp2 give dial tone
Logged
Stewart
Hero Member
*****
Posts: 1125


« Reply #3 on: June 28, 2012, 01:59:19 am »

1:

For SP1, set CallForwardOnNoAnswerNumber to e.g. SP2(1234) .  On your Asterisk system, set up 1234 as a feature code that routes to a DISA.  The DISA would answer, ask for a PIN (if desired) and then play dial tone.

Instead, if your SP1 provider supports two or more registrations on the same account, you could have both the OBi and Asterisk register to them.  Set up the Asterisk inbound route to delay answering, then send the call to DISA.  This method does not route DISA calls through the OBi, so it should be more reliable and have better quality.

2: If your SP1 provider supports multiple registrations as above, or if they permit unregistered (though authenticated) outgoing calls, simply set up a trunk on Asterisk to route the desired calls via your SP1 provider.  This does not involve the OBi at all.  If this solution is not feasible, please describe the constraints and I'll try to present an alternative.

What Asterisk distribution are you running (FreePBX, etc.)?

Who is your SP1 service provider?

If you have no connection to the PSTN, the OBi100 (instead of OBi110) should work for your application.
Logged
Pages: [1]
  Print  
 
Jump to:  

Powered by SMF 1.1.11 | SMF © 2006-2009, Simple Machines LLC