Dialplan help!

[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/networkpatterns.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]

hi mhank,
Paste your current dialplan rules here.

[mhank]

I have no dial plan at the moment.

I need help to how I should build it

[Tata]

Have you read the tutorial?
http://www.brekeke-sip.com/download/bss/v2_x/bss_tutorial_dialplan_en.pdf

You may find an example in the tutorial.
Have a fun.

[mhank]

Yes, I saw that one
but did not help me

but, I really somebody to "show me"

[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]

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

What would you suggest??

[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]

Try the example 10 in the DialPlan tutorial.
It seems you did not read it.

[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]

The best way to start with sip server is the quick start.
http://www.brekeke.com/products/products_sip_quickstart_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]

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]

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]

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.

[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!