[Asterisk-Users] RPID Issue

[Olle E Johansson]

Ray Van Dolson wrote:
> On Sat, Dec 31, 2005 at 10:05:19AM +0100, Olle E Johansson wrote:
> 
>>We're currently planning a new generation of chan_sip that will have a 
>>different authentication scheme, not based on the from: header unless 
>>it's a local policy to require the From: header to be the same as the 
>>Digest auth user name.
>>
>>So to summarize: The Sipura is doing the right thing, but Asterisk can 
>>not handle it today, since Asterisk requires a From: user name. You need 
>>to disable the caller ID in Asterisk, not in the Sipura.
> 
> 
> Gotcha.  Is there an open bug on this yet?  Or should their not be one since
> it is a planned feature for the future?  I'll just continue using my ghetto
> patch that uses RPID for authentication info as this "works" in our
> environment.

It's not really a bug, but an effect of the current architecture that is 
documented
and, well, there. Sorry. Will be fixed in a new architecture.

> Next RPID issue.
> 
> Our Asterisk server talks to our VoIP provider via a MediaCodes SIP gateway
> of some sort.  They also send us RPID headers.  Unfortuantely, in a format
> that Asterisk does not appear to understand:
> 
> <sip:5305715515 at 216.229.127.55>;party=called;npi=1;ton=2, <sip:5306802843 at 216.229.127.55>;party=calling;privacy=off;screen=yes;screen-ind=3;npi=1;ton=2
> 
> As you can see it's giving us the called party info first and the calling
> party info second.
> 
> get_rpid_num() appears to just check for the first ':' and grab the number
> immediately afterwards.  This is resulting in caller id being set to the
> called number, which really confuses customers obviously :-)
> 
> I'm guessing the above is an RFC compliant RPID header and Asterisk's
> behavior should handle it?
> 
> I hacked up another patch to address this:
> 
> http://webdev.digitalpath.net/~rayvd/dist/asterisk/rpid_multiple.patch
> 
> This works fine as long as we assume that only two entries can be present in
> the RPID header...
> 
Please submit that patch to the issue tracker at bugs.digium.com.

Thank you for contributing to Asterisk!

/O