#184 Call log integration and rewriting rules

[Anonymous]

Originally created by: r3gis...@gmail.com
Originally owned by: r3gis...@gmail.com

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

[Anonymous]

Originally posted by: r3gis...@gmail.com

Issue 228 has been merged into this issue.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Issue 489 has been merged into this issue.

[Anonymous]

Originally posted by: prrv...@gmail.com

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.

[Anonymous]

Originally posted by: r3gis...@gmail.com

> 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.

[Anonymous]

Originally posted by: SimonUmb...@gmail.com

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 ;-)

[Anonymous]

Originally posted by: lailo...@gmail.com

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.

[Anonymous]

Originally posted by: chri...@tu-chemnitz.eu

How about logging the unfiltered number?

[Anonymous]

Originally posted by: r3gis...@gmail.com

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)

[Anonymous]

Originally posted by: lailo...@gmail.com

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.

[Anonymous]

Originally posted by: crestani...@gmail.com

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