[Spce-user] Send peer inbound call straight to voicemail

[John Murray]

Jon,

The problem initially to be authentication.

The PBX is actually a gateway and does not register with SPCE. I need to give each gateway subscriber an SPCE subscriber so I use the trusted source model.
IP address of gateway + userpart of From: field matches the subscriber for them to make calls.
For calls to the subscriber I have a static contact/location pointing at the gateway.

However when a call is put out to the subscriber on the gateway and they are not available a call comes in from the gateway to SPCE with 'From:' the original caller 'To:' the voicemail number and 'Divert:' the subscriber details.

This call is rejected with '407 Proxy Authentication Required' as there is no PSTN peer and no trusted source matching the 'From:' because the From: is the originating caller in the PSTN.

I cannot add a peer with the gateway's address as in the logic this is checked before trusted sources and non of my subscribers are then matched.

Would it not be better to check trusted sources first?

Many thanks for your help.

Regards

John

-----Original Message-----
From: spce-user-bounces at lists.sipwise.com [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Jon Bonilla (Manwe)
Sent: 15 March 2013 09:27
To: spce-user at lists.sipwise.com
Subject: Re: [Spce-user] Send peer inbound call straight to voicemail

El Thu, 7 Mar 2013 22:09:44 -0000
"John Murray" <john.murray at skyracktelecom.com> escribió:

> Hi,
> 
>  
> 
> I want to send a specific call type to voicemail.
> 
>  
> 
> I send a call out to a PBX peer who has no voicemail. If the handset 
> is not answered I get a call from the peer with RURI 500 which is the 
> voicemail number, From is the original caller and a Divert: header 
> with the subscriber ddi.
> 
>  
> 
> I know how to take off the Divert header and get the subscriber ddi 
> but what would I put in the RURI to send straight to the subscribers voicemail?
> 
> I've tried prefixing 'vmu' to the ddi but that didn't work.
> 
>  

what was the ngrep-sip (including port 5062), the kamailio-proxy log and the asterisk console output of that call?