[Serusers] Getting rid of SIP URI in Caller ID

[Charles Ulrich]

On Friday 06 July 2007 04:41, Jiri Kuthan wrote:
> Hi Charles,
>
> Is the 'voip edition' somewhere available? I would be eager to give
> it a try on my WRT too.

Sure. There's no direct link, so you have to click through the following 
trail of links:

* Go to www.dd-wrt.com
* Downloads
* stable
* dd-wrt.v23 SP2
* voip
* dd-wrt.v23_voip.bin

Use this if you have a WRT54Gv4 or WRT54GL, otherwise check the 
documentation to find out which firmware image you need. DD-WRT works 
on a couple of other router brands too, such as Buffalo, Belkin, and 
ASUS.

> I'm a bit sceptical how much we can do about it, since the URIs are
> part of the SIP protocol machinery and it is up to discretion of a
> telephone implementation to show what it wants to show (there is no
> standard for what a telephone shall display).

You mean SIP doesn't have some kind of caller ID header? When these 
phones are registered directly with Asterisk, caller ID works exactly 
like a legacy non-SIP phone: Only a name and a number (or extension) 
appear on the display on an incoming call. Using SER as a proxy, we get 
the SIP URI instead of simply the telephone number.

> IMO you are then left with experimenting and changing SIP requests in
> a way that increases the chance that the phone shows what you want to
> be shown. There is no guarantee however it will work for all phones
> in a consistent way.
>
> If I were you, I would try appending P-asserted-identity or
> Remote-Party-ID header fields with tel URIs (benefit: use of a header
> field does not change request URI, which might have side effects
> otherwise, and use of TEL URI eliminates the domain). If that does
> not work, I would try to put TEL URI in request-URI.

I'll see what I can come up with but it looks like using SER as an 
outbound proxy isn't quite solving the main issue (NAT traversal) 
anyway.

Thanks!
-- 
Charles Ulrich
Ideal Solution, LLC -- http://www.idealso.com