olleb3 / csipsimple

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Call log integration and rewriting rules #184

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
When calls that has been rewritten are logged into the native logs should be 
un-rewritten ...

Tricky to do, but that would be the correct behavior

Original issue reported on code.google.com by r3gis...@gmail.com on 31 Aug 2010 at 9:34

GoogleCodeExporter commented 9 years ago
Issue 228 has been merged into this issue.

Original comment by r3gis...@gmail.com on 17 Oct 2010 at 10:54

GoogleCodeExporter commented 9 years ago
Issue 489 has been merged into this issue.

Original comment by r3gis...@gmail.com on 12 Dec 2010 at 4:25

GoogleCodeExporter commented 9 years ago
Thank for your answer

This problem only affects outgoing call that has a rewrite rule in filter (I 
don't think the rewrite rule applies to incoming calls?)

In fact I am obliged to prefix numbers with 0 because I use Keyyo -> Thank lot 
for the Wizard :)

In the rewrite rule, a check box that allows you to log in the history call the 
phone number before rewriting, would be welcome.

Original comment by prrv...@gmail.com on 12 Dec 2010 at 10:25

GoogleCodeExporter commented 9 years ago
> I don't think the rewrite rule applies to incoming calls?

Would be cool however to be able to use rewrite rules to reverse before writing 
in logs. Cause for incoming calls you have numbers with a leading 0. 
If it could be removed from logs could be really great :)

However, yes if I add something for outgoing calls it will even be automatic (I 
don't see any use case where users would deactivate this if available).
But I'd like to do something independent of incoming / outgoing, it would be 
really better in term of user experience ;).

P.S. : CSipSimple have a good Keyyo support, all the more so as now there is a 
Keyyo-VoIP app (under GPLv3) based on CSipSimple ;). But you can continue using 
CSipSimple, it's the same app finally, just re-branded by Keyyo.

Original comment by r3gis...@gmail.com on 12 Dec 2010 at 10:54

GoogleCodeExporter commented 9 years ago
Hi,

Have there been any advances regarding this issue? I'm using rewrite rules to 
dial into conference call numbers, pause and then provide my meeting id 
followed by #.
Currently, what I'm seeing in the logs is just the dial-in number, not the 
meeting id i dialed into, which would be much more useful to me :-).

Could you please give a quick update if this is on the timeline? I'd also not 
be adverse to coding a little myself ;-)

Original comment by SimonUmb...@gmail.com on 31 Jan 2013 at 10:07

GoogleCodeExporter commented 9 years ago
This is a vital feature to get working. I have no other dialers on my tablet, 
and having to install another (from what I understand) to get the number 
rewriting to work is not a viable solution.

I have a large company-maintained phone book, some numbers with prefixes and 
some without. My SIP provider does not accept any non-qualifying 7 or 10 digit 
numbers since they say this can be done on most handsets (as the handset best 
knows the area in which you are in), so I am at a loss.

Any ETA on this? It has literally been years.

The feature (filters) is basically already there, just not working IMO.

Original comment by lailo...@gmail.com on 13 Feb 2014 at 2:13

GoogleCodeExporter commented 9 years ago
How about logging the unfiltered number?

Original comment by chri...@tu-chemnitz.eu on 2 Apr 2014 at 10:56

GoogleCodeExporter commented 9 years ago
Not sure that we are all talking about the same feature here :). Maybe we 
should split this issue.
But before, to anyone, the csipsimple dialer has now something available to 
enable rewritings inside the dialer (press account button and click the checbox 
to apply rewriting). Used in combination with the long press + dial on call log 
row, it can maybe help some situations I think that are the concern of some 
comments here.

Else about storing unfiltered numbers it is not possible if we talk about the 
same thing. The point of this issue is :
We have an incoming or an outgoing or a missed or whatever (keep in mind that 
it might be dialed already rewritten by user from csipsimple dialer) call. 
Inside csipsimple call log list... Well no problem to log the actual sip uri. 
This one is by construction reachable using sip protocol as we already did a 
sip call using it. So it's OK for this part.
Now when csipsimple call logs integration is enabled, csipsimple tries to add 
an entry in android callog list. It tries to deduce a phone number from the sip 
uri. (If it's detected that there is not only digits will do nothing). That's 
fine in most case. But there is a failing scenario where it introduce in your 
android callog a number that is not reachable over gsm. It's when your sip 
provider use prefix for pstn gateway. In this case to get the gsm number from 
sip uri there is a transformation to do. And normally, this transformation is 
the exact reverse of one of the rewriting rules user of this provider might 
have already added.
In the example : you have a rewrite rule to add 0 for your provider to call 
over landline network. When you get a call from landline network it will have 
an extra 0 and would like it to be logged in android call logs without this 
extra 0. 
Sounds easy to do in this sample, but csipsimple rewriting rules allows regular 
expression and rversing a (multiple) regular expression is not a simple thing. 
At least for now I didn't find a way to do for all case. If somebody knows... 
Tell me :)

Of course, in the very special case of an outgoing call rewritten by 
csipsimple, yes there is something very simple that could be done. But it's a 
special case and adding such a hack will not solve cleanly the problem. Will 
remain incoming call problems and calls made directly from csipsimple dialer 
with user dialing something already written. (In the example with the extra 0 
already added)

Original comment by r3gis...@gmail.com on 2 Apr 2014 at 10:34

GoogleCodeExporter commented 9 years ago
Thanks for the rewriting rules... all I needed. Yup, it would have been better 
to split this issue into two from the start. So much unwanted traffic.

Original comment by lailo...@gmail.com on 3 Apr 2014 at 12:59

GoogleCodeExporter commented 9 years ago
Is it possible to get ALL the calls (gsm + SIP) done with csipsimple client in 
the internal cSipSimple call log? 

Original comment by crestani...@gmail.com on 13 Apr 2015 at 1:11

GoogleCodeExporter commented 9 years ago

Original comment by r3gis...@gmail.com on 22 Jun 2015 at 11:30