On Wed, Aug 20, 2008 at 10:44 AM, Damian Krzeminski <[EMAIL PROTECTED]> wrote:
> Scott Lawrence wrote:
>> On Tue, 2008-08-19 at 18:23 -0400, M. Ranganathan wrote:
>>> Hello
>>>
>>> I have to do a test for AT&T where the Display Name set on the phone
>>> is forwarded with the outbound Invite to the ITSP. However, I notice
>>> that sipXproxy seems to drop the display name when forwarding.
>>> Attached is a merged and filtered trace (I filtered the relevant
>>> sections using sipx-trace ).
>>>
>>> Notice that in Frame 17 the INVITE has the display name "user six" in
>>> the From header. However notice that in Frame 19, the display name is
>>> dropped.  Not sure if I should I be setting the display name
>>> elsewhere. If this is a possible bug, I can file a report.
>>
>> The display name was removed because you had configured a caller alias.
>> You can tell because the Record-Route header has a fromalias attribute
>> in it - this can be decoded using the sipx-readroute script:
>>
>> sipx-readroute
>>
>> Record-Route: 
>> <sip:192.168.5.240:5060;lr;sipXecs-rs=%2Afrom%7EZHJqdzZvcnB6bQ%60%60.400_authrules%2Aauth%7E.700_fromalias%2Aalias%7EPHNpcDo3MzItMzIwLTQyODBAZXhhbXBsZS5jb20%27O3RhZz1kcmp3Nm9ycHpt.700_fromalias%2Acaller%7EInVzZXIgc2l4IiA8c2lwOnVzZXI2QGV4YW1wbGUuY29tPjt0YWc9ZHJqdzZvcnB6bQ%60%60%21c7c2d612599987c01f743df2b1f6a0e3>
>>     *from=drjw6orpzm
>>     400_authrules*auth=
>>     700_fromalias*alias=<sip:[EMAIL PROTECTED]>;tag=drjw6orpzm
>>     700_fromalias*caller="user six" <sip:[EMAIL PROTECTED]>;tag=drjw6orpzm
>>
>> The 'caller' attribute is the original caller url, and the 'alias'
>> attribute is the value it is replaced by.
>>
>> The CallerAlias authplugin takes the value from its IMDB table and uses
>> it as a full url, replacing only the 'tag' value.
>>
>> Apparently, sipXconfig won't let you enter a full URL value in the
>> alias, but if it would, then you could control any part of the url sent
>> in the aliased call except the From tag.
>>
>>
>>
>
> Not sure why sipXconfig has this limitation. But if it's just an overly
> eager error checking that should be easy to remove.

There must be an error check. It must be a grammatically valid SIP URI.

A precise error check can be applied by using the JAIN-SIP factory to
parse the entered URI.

I do think it would be easier for the user to specify independent
fields for display name, user name and domain rather than expect them
to enter a SIP URI in a grammatically correct fashion.

URI parameters are not relevant for for From headers. ( I dont think
they are allowed per the RFC but I need to check that ).

Ranga



> There is a JIRA issue on it:
> http://track.sipfoundry.org/browse/XCF-2633
> D.
>
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-- 
M. Ranganathan
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