Closed GoogleCodeExporter closed 9 years ago
Actually the behavior is :
If caller is DisplayName <sip:xxxxx@domain.name>
=> If xxxxxx is not a phone number :
-> Display "DisplayName" if present else display xxxxx
=> If xxxxxx is a phone number :
-> Search if this phone number exists in the contact
-> If in contact display username from contact
-> Else display phone number
My understand of your request is change the last point from display phone
number to display "DisplayName" if present else display phone number?
I'm right?
Original comment by r3gis...@gmail.com
on 24 Jul 2010 at 8:34
Mhhh written too fast.
Actually last point does what is proposed to solve your issue.
If you know exactly what sip id is sent by caller / callee, it can help me to
understand what is wrong with my parsing rules. But should be ok, just
re-tested it.
Original comment by r3gis...@gmail.com
on 24 Jul 2010 at 8:41
I set in asterisk a display name "DisplayName" and a number. The Android
clients shows only the number. On a Snom IP phone "DisplayName" appears.
I think in SIP the "From:" Header is "DisplayName <sip:1234567890@10.20.30.40>".
Original comment by mathias4...@gmail.com
on 24 Jul 2010 at 10:29
I've just added some unitary test for this case.
If From header is "DisplayName <sip:1234567890@10.20.30.40>" then "DisplayName
" is displayed in the in call activity. Unless 1234567890 exists in your
android contacts and then contact name is displayed.
It is also what I observed when using a sip softphone on my PC, setting my
caller id and place a call to my android handset : display name is properly
displayed.
To check that csipsimple receive the good contact uri, you can install the
latest build :
http://code.google.com/p/csipsimple/downloads/detail?name=CSipSimple_0.00-12-06.
apk (uninstall previously installed version before installing this one).
There is a new feature : the (i)nfo button that show you statistics about the
current call, and that include the "From " field from SIP.
Besides, this version will allow you to send me more detailed logs so I can see
exactly what is announced by your sip server. (Use aLogcat application for
example to send me your logs just after you reproduced an invalid scenario) .
Original comment by r3gis...@gmail.com
on 25 Jul 2010 at 9:16
OK, now I see the name within the call. Only in the missed calls list is no
name.
Thanks a lot for your great work.
Original comment by mathias4...@gmail.com
on 25 Jul 2010 at 1:29
Ok, that's true, it's not fully implemented in CallLog list.
I change the issue title to track this task.
Original comment by r3gis...@gmail.com
on 25 Jul 2010 at 2:07
It would be usefull to lookup in the callerid(num) not name !
For example Ippi adds "33" and delete the firtst 0 before the phone number and
sends it as the callerid name.
As it is just "33" and not "0033" nor "+33" nothing can match ...
I also prefix the callerid name with a tag (*) when it's a personnal call from
another voip provider ... Once again nothing can match...
Anyway the callerid number is always right and can be checked if nothing good
is found in the callerid name !
Don't hesitate to ask, I will follow
Original comment by cedricli...@gmail.com
on 30 Jul 2010 at 5:08
[deleted comment]
Now (0.00-15-05) call logs add their cached_numbers. So should appear with the
good display name in call logs (native & csipsimple).
As for reverse matching (your last comment) : see issue 184.
Still have to be fixed... and somehow complicated to do automatically.
Original comment by r3gis...@gmail.com
on 25 Oct 2010 at 11:11
Original issue reported on code.google.com by
mathias4...@gmail.com
on 24 Jul 2010 at 7:55