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Author Topic: Dial plan explanation  (Read 2091186 times)
RonR
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« Reply #80 on: February 02, 2012, 08:15:24 pm »

Physical Interfaces -> LINE Port -> InboundCallRoute : {SP1(12341234567)}

Voice Services -> SP1 Service -> X_InboundCallRoute : {12341234567:li},{aa}

Voice Services -> Auto Attendant -> UsePIN : (checked)

Voice Services -> Auto Attendant -> PIN1 : (numeric PIN)

where 12341234567 is US cell_phone1.
« Last Edit: February 04, 2012, 01:15:58 pm by RonR » Logged
Stewart
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« Reply #81 on: February 02, 2012, 10:12:56 pm »

Incoming call to Obi's PSTN port have to be transferred to my US cell_phone1, without pause or PIN access just blind transfer to my cellphone ...
Though RonR's solution will do exactly what you asked for, it seems like a very unusual request, because your friend would be unable to receive calls at home!  Is this an "extra" line that she doesn't presently use, e.g. it's bundled with her Internet and TV service, but she takes all calls on her cell phone?  Otherwise, please post details, as there may be a better way to accomplish your goals.
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Albert
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« Reply #82 on: February 04, 2012, 09:33:23 am »

Dear.
I used config from RonR and everything Great.!!.
Only inbound call from PSTN to SP1 no blind forward.
I used:
Physical Interfaces> LINE> InboundCallRoute> SP1(US cell_phone1)
After 7 ring call inbound throw PSTN finaly bridge to US cell_phone1 but call hang-hook continued a unknown number ring and hang-hook again
What could be wrong?
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RonR
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« Reply #83 on: February 04, 2012, 01:01:42 pm »

After 7 ring call inbound throw PSTN finaly bridge to US cell_phone1 but call hang-hook continued a unknown number ring and hang-hook again

I don't know what this means.  I'm not familiar with the term 'hang-hook'.


In my previous example, I did leave out the curly-braces that should be around the rule:

Physical Interfaces -> LINE Port -> InboundCallRoute : {SP1(12341234567)}

should work.  I use that here to forward incoming landline calls to my cell phone.

-----

Upon further review, either should work equally well:


Physical Interfaces -> LINE Port -> InboundCallRoute : SP1(12341234567)

or

Physical Interfaces -> LINE Port -> InboundCallRoute : {SP1(12341234567)}


Curly-braces are not required when there is only a single rule.
« Last Edit: February 04, 2012, 01:32:01 pm by RonR » Logged
Albert
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« Reply #84 on: February 06, 2012, 08:59:25 am »

Dear
I can't forward Incomng PSTN call to Google Voice
I've configured OBI for Google Voice (SP1), and make outgoing calls, blind forward from GV to Line port with no problem. Everything is great.!!
SP1 Services> X_InboundCallRoute> {cell_phone:li},{aa}
When Inbound call is connected throw PSTN port, it can't forward to Google Voice (SP1). The phone (PSTN) continues to ring and not forward to GV
Line Port> InboundCallRoute> SP1(cell_phone)
Somebody have any idea why not forward?
Regards
Albert
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Stewart
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« Reply #85 on: February 06, 2012, 09:21:51 am »

What, if anything, shows in Call History after a failing call?

If you have one or more phones connected directly to the PSTN line (as opposed to being connected to the OBi's PHONE port), it's normal for them to ring while the forward is being set up, and until the cell phone is answered.
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Albert
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« Reply #86 on: February 06, 2012, 03:13:14 pm »

Dear Stewart
Yes, the call history after failing call show the caller number. I dont have any phone connected directly to PSTN line. When somebody call me throw my PSTN line port, it can't forward to Google Voice (SP1). The caller's phone continue ring and not forward to GV
Line Port> InboundCallRoute> SP1(my_cell_phone)
Regards
Albert
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Stewart
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« Reply #87 on: February 06, 2012, 03:35:00 pm »

If you manually dial your cell phone number from the phone connected to the OBi (using GV), does the call work as expected?  If not, troubleshoot that first.

If the manual call works ok, try another forwarded call.  Allow at least 15 seconds (after the PSTN caller hears ringing) for your cell phone to ring.  Then, log into the OBi and go to Status -> Call History.  You should see several lines of information for the call in question.  Confirm that the Peer Number for the Outbound side is correct.  Then, post everything in the History for that call (mask the actual phone numbers).

What country / city is the OBi in?
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Dogzipp
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« Reply #88 on: February 07, 2012, 06:21:31 pm »

Can anyone help a little. I am overwhelmed with the amount of options the Obi has, and I am not sure as to where exactly put things..

I want to use my LINE mainly for all calls, except:

Calls that start with 4 (8 digits) to go through SP1
Calls that start with 1 (11 digits) to go through SP2 (Google Voice).
Everything else through the PSTN (911, 1xxx and 8 digit calls that start with 2-3 and 5-8)

I was thinking about this:

Physical Interfaces -> Phone Port -> Primary Line : PSTN Line
Physical Interfaces -> Line Port -> Digitmap : (<**1>4xxxxxxxS0|<**2>1xxxxxxxxxxS0|xx.)

But I am not 100% sure.. I see some people do the digits maps directly on ITSP Profile A, but I guess that's when your main line is SP1?

I would appreciate any configuration help, on how to make things work.
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RonR
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« Reply #89 on: February 07, 2012, 06:31:38 pm »

I would suggest:

Physical Interfaces -> PHONE Port -> PrimaryLine : PSTN Line

Physical Interfaces -> LINE Port -> DigitMap : (<**1>4xxxxxxx|<**2>1xxxxxxxxxx|911|1xxx|[2-35-8]xxxxxxx)

Each of these rules is an exact match, therefore no S0 is needed anywhere.
« Last Edit: February 07, 2012, 07:01:31 pm by RonR » Logged
Dogzipp
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« Reply #90 on: February 07, 2012, 06:56:35 pm »

Each of these rule is an exact match, therefore no S0 is needed anywhere

What does the S0 do exactly?
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RonR
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« Reply #91 on: February 07, 2012, 07:00:49 pm »

What does the S0 do exactly?

Sn overrides the default interdigit timer.  S0 would force an immediate exit of the Digit Map Processor if the asociated rule mates but there is sill an indefinite match rule possible.
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Dogzipp
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« Reply #92 on: February 07, 2012, 07:25:09 pm »

Wouldn't <**2>1xxxxxxxxxx and 1xxx conflict then?

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RonR
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« Reply #93 on: February 07, 2012, 08:11:16 pm »

You're right.  Unless there's something unique between those two that you can use to qualify them, you'll have to add an Sn (1xxxS2) to allow possibly dialing a fifth digit after 1xxx.
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Dogzipp
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« Reply #94 on: February 07, 2012, 08:34:03 pm »

You're right.  Unless there's something unique between those two that you can use to qualify them, you'll have to add an Sn (1xxxS2) to allow possibly dialing a fifth digit after 1xxx.


So (<**1>4xxxxxxx|<**2>1xxxxxxxxxx|911|1xxxS2|[2-35-8]xxxxxxx) should be allright?

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RonR
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« Reply #95 on: February 07, 2012, 08:58:09 pm »

Looks good, but the real test is to plug it in and try the various patterns to make sure they do what you want.

The Call History will show you the outcome of each test.
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Lowdough
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« Reply #96 on: February 09, 2012, 03:23:17 pm »

I'm using the OBI on Callcentric (port 2, the default port) and it's working great.  Port 1 is Google voice and it's working great, too.

But I'm a cheapskate and the idea of paying Callcentric for 800- number calls just doesn't sit well, so I'd like to modify the digit map to peer to SIP Broker (**275) before sending out the toll-free digit strings.  When I was using a regular old SIP, I used the dial plan below, but I have to admit to having a hard time understanding the multi-layer digit maps the OBI's use.  Anyone know how to convert this to a digit map to accomplish that?

(*xx.|*xxx|*75xx|[3469]11|0|00|<:**275*>1800x.|<:**275*>1866x.|<:**275*>1877x.|<:**275*>1888x.|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|**275*x.)

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RonR
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« Reply #97 on: February 09, 2012, 03:43:46 pm »

This will send your toll free calls though SP2 (Callcentric) via Sip Broker instead.

Service Providers -> ITSP Profile B -> General -> DigitMap:

(*123|**275*xx.|[2-9]11|<**275*>(18(00|88|77|66|55)xxxxxxx|<1>8(00|88|77|66|55)xxxxxxx)|
1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1aaa>[2-9]xxxxxx|011xx.|(Mipd)|[^*#]@@.'@'@@.)

where aaa is your local area code.
« Last Edit: February 09, 2012, 04:52:19 pm by RonR » Logged
Gennady
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« Reply #98 on: February 20, 2012, 09:49:06 am »

I have PSTN line in Canada and my VOIP provider located in Ukraine and mapped to SP2.  I'm trying to setup the rule: prefix 8044 going to SP2 and 8044 substracted from #. Here is my settings:
Phone Port ->OutboundCallRoute:
{([1-9]x?*(Mpli)):pp},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<8044:>(Msp2)):sp2},{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}
ITSP Profile B -> DigitMap:
(<8044:>xxxxxxx|xxxxxxx)
LINE Port-> DigitMap:
(xxxxxxxS4|1xxxxxxxxxx|xx.)
Works Ok, but it's a long delay before phone start ringing. I was following Ron's suggestions and changed Line Digimap to
[2-9]11|416xxxxxxx|905xxxxxxx|647xxxxxxx|289xxxxxxx|1xxxxxxxxxx|3xxxxxx|011xxxxxxxxxxxx)
After this change all calls started with 8044 are not going to SP2 at all. Any help?



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RonR
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« Reply #99 on: February 20, 2012, 10:09:30 am »

Leave the PHONE Port DigitMap and OutboundCallRoute, and the ITSP Profile B DigitMap at Default.

Assuming your PrimaryLine is PSTN Line, set:

Physical Interfaces -> LINE Port -> DigitMap:

([2-9]11S0|[2-9]xxxxxx|1xxxxxxxxxx|011xx.|<8044:**2>xxxxxxx)
« Last Edit: February 20, 2012, 10:46:54 am by RonR » Logged
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