[Rtcomm] [Rtcomm] [Bug 4929] New: Direct (no proxy or registrar) SIP URI dialing

[From: bugzilla-daemon at maemo.org]

https://bugs.maemo.org/show_bug.cgi?id=4929

           Summary: Direct (no proxy or registrar) SIP URI dialing
           Product: Chat and Internet Call
           Version: 4.1.3 (5.2008.43-7)
          Platform: All
        OS/Version: Maemo
            Status: NEW
          Severity: enhancement
          Priority: Low
         Component: SIP
        AssignedTo: rtcomm at maemo.org
        ReportedBy: maemo at subs.maneos.org
         QAContact: sip-bugs at maemo.bugs


SOFTWARE VERSION:
5.2008.43-7

STEPS TO REPRODUCE THE PROBLEM:
1a. Start with no SIP accounts configured, or any configured ones disabled.
2a. Start Internet call application.

or 

1b. Start with no SIP accounts configured, or any configured ones disabled.
2b. Start browser.
3b. Open <https://bugs.maemo.org/attachment.cgi?id=1255>.
4b. Tap the sip: link.

EXPECTED OUTCOME:
a. Option to enter a SIP URI to be dialed.

or

b. Internet call application starts and attempts to dial the SIP URI using
NAPTR/SRV/A DNS lookups to locate the appropriate server.

ACTUAL OUTCOME:
"No accounts set up".

REPRODUCIBILITY:
Always.

OTHER COMMENTS:
This would be useful since many SIP services do not allow outgoing calls to
third-party domains.

One question is what to put in the From: header in such a case.  An "Anonymous"
(RFC3261 8.1.1.3 & 20.20) header would seem appropriate.


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