Dialplan help!

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mhank
Posts: 7
Joined: Thu Mar 22, 2012 1:39 am
Location: Denmark

Dialplan help!

Post by mhank »

1. Brekeke Product Name and version:
BRekeke SIP server ver. 3

2. Java version:

3. OS type and the version:
Windows 2008 server

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:

Any one who can help me with a dialplan
??

All inbound must go to the "matching" account

All outbound calls should be sent to our MEDIA PROXY at IP: xxx.xxx.xxx.xxx
taitan
Posts: 237
Joined: Sat Mar 15, 2008 10:39 pm

Post by taitan »

hi mhank,
Paste your current dialplan rules here.
mhank
Posts: 7
Joined: Thu Mar 22, 2012 1:39 am
Location: Denmark

Post by mhank »

I have no dial plan at the moment.

I need help to how I should build it
Tata
Posts: 223
Joined: Sun Jan 27, 2008 1:03 pm

Post by Tata »

Have you read the tutorial?
http://www.brekeke-sip.com/download/bss ... lan_en.pdf

You may find an example in the tutorial.
Have a fun.
mhank
Posts: 7
Joined: Thu Mar 22, 2012 1:39 am
Location: Denmark

Post by mhank »

Yes, I saw that one :)
but did not help me

but, I really somebody to "show me"
Tata
Posts: 223
Joined: Sun Jan 27, 2008 1:03 pm

Post by Tata »

> All inbound must go to the "matching" account
> All outbound calls should be sent to our MEDIA PROXY at IP: xxx.xxx.xxx.xxx

I can not suggest any dialplan because it is lack of information!
How do you decide "inbound" and "outbound" in your plan?
mhank
Posts: 7
Joined: Thu Mar 22, 2012 1:39 am
Location: Denmark

Post by mhank »

I have never worked with brekeke before - So I have no idea on how the system works!

What would you suggest??
mhank
Posts: 7
Joined: Thu Mar 22, 2012 1:39 am
Location: Denmark

Post by mhank »

all calls arrives from IP : (e.g.): 123.123.123.123

in format b_number@123.123.123.123

the b-number should be sent to the Matching account ID on the brekeke server


All outbound calls must be sent to IP: 123.123.123.123
Laurie
Posts: 245
Joined: Mon Jan 07, 2008 12:25 pm

Post by Laurie »

Try the example 10 in the DialPlan tutorial.
It seems you did not read it.
mhank
Posts: 7
Joined: Thu Mar 22, 2012 1:39 am
Location: Denmark

Post by mhank »

Hi,

sorry no - that does not help me

I think the way it should work, is very simple/clear, but maybe brekeke sip server, simply can not do it.

If it can work, I simply just need somebody to tell me how - As I am not able to make it work.

I would not ask the question in this forum, if I could make it work.
I kindly ask " can somebody please show me" or at least tell me "not possible" and then we can start to look at a other product

thanks.
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

The best way to start with sip server is the quick start.
http://www.brekeke.com/products/product ... simple.php

You do not need a dial plan to register two sip phones or one sip phone and one outside pbx to sip server and make calls to each other. You DO need dial plan if you want security and not allow anyone to access your sip server thru sip unless authorized, with the exeption of setting registration requirement in configuration. The forum is not the best place to learn the basics of Brekeke sip server.

I would suggest you consider the Brekeke PBX where security and many common dial plans are already included with the bundled sip server. This makes a much easier startup experience. I hope this helps.
mhank
Posts: 7
Joined: Thu Mar 22, 2012 1:39 am
Location: Denmark

Post by mhank »

let me tell you what I am looking for

we are Telco provider, who delivers:

Hosted PBX
MOBILE PBX
SIP TRUNKS

we are looking for a extra SIP TRUNK SERVER

(a server where customers with their OWN PBXs can use to send & receive calls"

IS brekeke SIP server the correct product for that??
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

Yes.

The advanced SIP server is a carrier class sip proxy. Should handle your requirements. But, configuration is required. There are connection issues that need to be addressed like type of authentication such as login or ip authorized for security and directing calls to those end points that are login authed vs ip authed. At a minimum to set this up would require a good understanding of the sip server dialplan tutorial and following the quick setup to start. Brekeke's strength is in the ability to manipulate the sip headers both coming and going from the sip server to insure you can achieve the desired connection.

To help describe a dial plan to you think of it this way.

SIP packets come in from a caller and pass thru the dial plan. It looks at the first rule and if the matching pattern in the cial plan matches what is in the sip incoming header it executes the deploy pattern, if not it looks to the next dial plan. The order you put them in is important because as a soon as a match is made the deploy fires and doesn't look to any more dialplans. You can try to match the "to", the "from" the ip address, the phone number or anything.
Tata
Posts: 223
Joined: Sun Jan 27, 2008 1:03 pm

Post by Tata »

mhank,

Try this dialplan rule which will meet your requirement.
---------------------
Matching Patterns:
$request = ^INVITE
$addr = 123.123.123.123
To = sip:(.+)@

Deploy Patterns:
To = sip:%1@<MEDIA_PROXY_IP_Address>
---------------------

If you can not understand it, you need to learn about SIP, Regex and DialPlan.
User avatar
KentC
Posts: 108
Joined: Fri Dec 09, 2011 2:27 pm
Location: rw-rw-rw-

Post by KentC »

Let me jump in with my novice knowledge :)

Matching Patterns:

always start with this

$request=^INVITE (Must have)

Then add:
From=sip:(.*)@ (Any expected ANI,this receives whatever as anything inside this is a phone number/DID)

like... From=sip:(variable)@ <- (brackets make it a variable)

To=sip:.1(999999999)@ (substitute for DNIS,that's american phone # with the +1 outside the variable then phone number, which is the number you are receiving) *where you are trying to send the phone call



Deploy Patterns:

From=sip:(%1)@brekeke's IP (variable one in parenthesis)

Then

To=sip:(%2)@customer's IP (variable two and the @IP of where it's leaving off towards like if its going to a cisco router or customers address

Recap:

Matching Patterns:
$request=^INVITE
From=sip:(.*)@
To=sip:.1(999999999)@

Deploy Patterns:
From=sip:(%1)@brekeke's IP
To=sip:(%2)@customer's IP

Hope that helps some, I know how hard it is to learn this stuff. My only sip experience is from Brekeke!
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