2 Replies Last post: Oct 30, 2007 1:11 AM by Guest  
Guest

Oct 17, 2007 6:33 AM

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

Guest
1. Jul 2, 2009 7:18 AM in response to: Guest
[OpenSIPStack] [OpenSBC] Sip Registration to Voip Provider.
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

[d-1100]
  • 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----------------------------------


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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Guest
2. Oct 30, 2007 1:11 AM in response to: Guest
Re: [OpenSIPStack] [OpenSBC] General Questions
Hi Joegen

I have several questions but first let me discribe my network

I have installed sipXecs box with Centos5 DHCP, DNS and NTP
static192.168.1.2 .
Gateway 192.168.1.1
netmask 255.255.0.0

I have installed openSBC box with two NIC's one with an
1st NIC
static 192.168.1.3 c
Gateway 192.168.1.1
netmask 255.255.0.0

2nd NIC
static 69.x.x.241 public
gateway 69.x.x.1
netmask 255.255.255.0

I am able to remotely config openSBC over the Internet and from the private
netwrok from the openSBC box configure the sipXecs box

I have set the openSBC mode to B2BUpperReg.
Static RTP Media address to 192.168.1.2
Now openSBC should bind it's listeners to both NIC's.

Additionally i have these settings

in B2BUA i have
[sip:*@:sipx.domain.net:*] sip:192.168.1.2:5060 which is pointing to a user
account on the sipX

Relay Routes
[sip:*@:sipx.domain.net:*] sip:192.168.1.2:5060

Internal DNS Mapping
[sip:*@:sipx.domain.net:*] sip:192.168.1.2:5060

Upper Registration Router
[sip:sipx.domain.net*] sip:192.168.1.2:5060

domain.net is a placeholder for a read domain.

The goal is to have openSBC handoff to sipX and vise-versa to handel the
sipX proxy services. I want sipX to register external Users, phones and
dialplans. Does the B2BUA, Relay routes, DNS Mapping and Upper registration
Routes set properly to get the job done? Have I address everything needed?

The remote phones TFTP are pointed to sipX as well as their OutBoundProxy
plus there phone numbers username and password. Internally the sipX gateway
setting is pointed to the openSBC box.

Maybe the questions & answers could be posted to the wiki in the future.

r



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