Re: [OpenSIPStack] OSBC <==> sipX Send-Reg-NO

[<jo...@op...>]

Ok lets use more descriptive domain naming.

Outbound calls flow = UA1 ->  SIPX -> OSBC -> Provider-GW
UA Dialstring =  sip:121...@si...
SipX = sipx01.mydomain.com.
OpenSBC = osbc01.mydomain.com
Provider = gw01.myprovider.com

In this sample, The call will reach OpenSBC with the following INVITE

    INVITE sip:121...@os... SIP/UDP
    From: "foo"<sip:fo...@si...>;tag=123
    To: sip:121...@si...


B2BUA-Route =  [sip:*mydomain.com*;from=gw01.myprovider.com] 
sip:gw01.myprovider.com

    INVITE sip:121...@si... SIP/UDP
    From: "foo"<sip:fo...@si...>;tag=123
    To: sip:121...@si...


Now just reverse this process for you egress traffic and you are all set.


voice wrote:
> Hi Joegen
>
> [sip:*@yourinternal-domain*;from=provider-domain]
> sip:your-internal-domain.com
>
> provider-domain == Termination
> yourinternal-domain == OSBC
> your-internal0domain == sipX
> xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
> provider-domain == Origination
> yourinternal-domain == sipX
> your-internal0domain == OSBC
>
> Is this what you mean?
>
> r
>
> ----- Original Message ----- 
> From: "jo...@op..." <joe...@gm...>
> To: "voice" <vo...@ne...>;
> <ope...@li...>
> Sent: Sunday, December 07, 2008 7:02 PM
> Subject: Re: [OpenSIPStack] OSBC <==> sipX Send-Reg-NO
>
>
>   
>> If you do not need to authenticate with your trunk provider, then you
>> not need to use SIP Trunking Module to place your calls.  You may simply
>> treat your Provider as a normal PSTN Gateway.   Get rid of you SIP-Trunk
>> config and use B2BUA-Routes instead.  I have given an example of a route
>> below.
>>
>> Rewrite-To-URI=  true
>> Route=  [sip:*@yourinternal-domain*;from=provider-domain]
>> sip:your-internal-domain.com
>>
>> - the from parameter in the filter will become the from domain in the
>> outbound invite-voice wrote:
>> - Rewrite-To-URI will force the To-URI to be rewritten based on the Route.
>>
>> Joegen
>>     
>>> Hi Joegen
>>>
>>> In the Sip-trunk config i have selected
>>>
>>> Send-Reg=NO
>>>
>>> I am using Broadvox wholesale, which means i have 2 SIP TRUNK'1
>>>
>>> One for Termination x.x.x.10
>>> One for Orgination    x.x.x.20
>>>
>>> I do not need username and password settings in <account> and maybe
>>> <Transient*> if i knew what Transient was used for*
>>>
>>> In light of these facts How should the Sip-Trunk Configure be setup or
>>> constructed and what is the bear minimum needed?
>>>
>>> r
>>>
>>>
>>>
>>>       
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