[OpenSIPStack] [OpenSBC] Sip Registration to Voip Provider.

[Joegen E. B. <joe...@gm...>]

This is a cross post.
---

Hi,

SIP Trunking is not prime time yet but you may already try it using the 
latest CVS copy of OpenSBC/OpenSIPStack.  To Enable Trunking, you must 
provide an XML configuration in "SIP Trunk Config".  Below is a template 
XML config.   In this sample config, sip:win32.opensipstack.org is 
assumed to be the internal domain of OpenSBC while 
sip:opteron.opensipstack.org is the domain of your SIP Provider.

[SIPTrunk]
* trunk-name:  This is the unique name OpenSBC will use to identify you 
SIP Trunk
* route-set:  This is the DNS resolvable domain or IP address of your 
trunk provider
* sip-domain:  This is the SIP Domain used as the host part of the To 
and From URIs
* expires:  Global expire interval for trunk registrations in seconds

[Trunk-Accounts]
* account - An instance of a virtual UA that will register to the Trunk 
Provider domain
** user-name - The user part of the From-URI
** auth-user-name - User name used for Authorization and Authentication
** auth-password - Password used for Authorization and Authentication
** inbound-route - URI specifying the identity of the UA in the internal 
domain
** expires - If set, this will be the expires used when the virtual UA 
registers to the Trunk Provider

[Transient-Accounts] - Transient accounts are similar to normal 
Trunk-Account in terms of the parameters.  The only difference is that 
they are also meant to be shared (in round robin fashion) by calls which 
are not defined in the normal trunk-accounts.  This is normally used if 
you have a few accounts with a Trunk Provider and is meant to be shared 
by all your external users.



------------------------START OF XML 
CONFIG----------------------------------

<root>
   <siptrunk trunk-name="opteron.opensipstack.org"
             route-set="opteron.opensipstack.org"
             sip-domain="opteron.opensipstack.org"
             expires="10">

      <trunk-accounts>
         <account user-name="1001"
                  auth-user-name="1001"
                  auth-password="1001"
                  inbound-route="sip:90...@wi..."
                  expires="3600" />
         <account user-name="1002"
                  auth-user-name="1002"
                  auth-password="1002"
                  inbound-route="sip:90...@wi..."
                  expires="3600" />
         <account user-name="1003"
                  auth-user-name="1003"
                  auth-password="1003"
                  inbound-route="sip:90...@wi..."
                  expires="3600" />
      </trunk-accounts>

      <transient-accounts>
         <account user-name="1001"
                  auth-user-name="1001"
                  auth-password="1001"
                  inbound-route="sip:90...@wi..." />
         <account user-name="1002"
                  auth-user-name="1002"
                  auth-password="1002"
                  inbound-route="sip:90...@wi..." />
         <account user-name="1003"
                  auth-user-name="1003"
                  auth-password="1003"
                  inbound-route="sip:90...@wi..." />
      </transient-accounts>

   </siptrunk>
</root>

Joegen

Dinesh Dialani wrote:
>
> Hi All,
>
>  
>
> I want to use Open SBC for *SIP TRUNKING*.
>
>  
>
> Here is the scenario.
>
>  
>
> Internal LAN                                          External LAN
>
> ------------------------------------
>
>                                     |
>
> Softphone ----> PBX ------|----> OpenSBC -----> Voip Provider
>
>                                     
> |                                              
>
>                                     |
>
> -------------------------------------
>
>  
>
> I wish that only OpenSBC should be visible to external world and thus 
> it should be able to register itself to VoipProvider.
>
>  
>
> Also I want that OpenSBC should use Enum lookup first for E-164 
> numbers on our enum server and if OpenSBC does not receive any 
> response from Enum server then it should be able to connect the call 
> through Voip Provider.
>
>  
>
> Now here are the questions.
>
>  
>
>    1. How to register OpenSBC with VoipProvider irrespective of PBX?
>
> t means that there should be fields in Web GUI to enter registration 
> request for Voip Provider and the moment OpenSBC service is started, 
> it should register itself with Voip Provider.
>
>    2. How to set preference order between Enum lookup and normal call
>       through VoipProvider?
>
> It means whenever an Sip INVITE is sent from PBX to OpenSBC for long 
> distance calls, OpenSBC should first search that number in our enum 
> servers and if it is found, call is made directly to the Receiver else 
> OpenSBC should direct the call to VoipProvider and create a normal call.
>
>    3. What are the entries to be given on B2B routing page for above
>       Enum and normal call to VoipProvider?
>
>  
>
> Thanks in advance for you help.
>
>  
>
> Dinesh
>
>  
>
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