Summary:ASTERISK-16763: [regression] Caller ID in SIP PAI Header is anonymous instead of Number
Reporter:1stbs (1stbs)Labels:
Date Opened:2010-10-03 19:43:30Date Closed:2012-02-16 07:21:19.000-0600
Versions:Frequency of
Description:The current enviroment is used:
Openser as Callserver and registrar
Asterisk 1.8 rc2 (current dahdi-lunix & dahdi tools svn head) as Gateway to Avaya Integral over QSIG
If a call is etablished the information in PAI contains anonymous instead of the dialed number and the phones which support this PAI header shows that you are connected with anonymous.
it seems that the function int ast_party_id_presentation(const struct ast_party_id *id) is wrong.  i have changed the selection of the winning  presentation value from if (name_priority < number_priority) to if (name_priority > number_priority). Now it behaves correct in that scenario.


Asterisk 1.8 before Beta 3 has not this bug.
Comments:By: David Brillert (aragon) 2012-01-18 14:05:05.495-0600

Is this something that would be fixed by https://reviewboard.asterisk.org/r/1673/ ?
Patch is submitted in SVN

By: Joshua C. Colp (jcolp) 2012-01-18 14:10:34.394-0600

No. This issue is about an outgoing INVITE with PAI, not about parsing.

By: Joshua C. Colp (jcolp) 2012-01-18 15:29:55.968-0600

Unfortunately after looking at the code in question as well as the changes that occurred in the mentioned versions things appear to be working as they should. To proceed with this issue I believe I will need to have a log of the complete console output as well as a pri trace. Thanks.

By: Leif Madsen (lmadsen) 2012-02-16 07:21:11.399-0600

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines