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<Wizzup> sicelo: did they maybe change the dbus interface or something
<Wizzup> doesn't look like it
<Wizzup> looks like 2.1 has ussd fixes for quectel
<sicelo> yes dbus interface is unchanged
<Wizzup> I think we can probably rebase relatively easily then
<sicelo> they recently changed the ofono atoms stuff. i have not been able to get the org.ofono.RadioSettings interface to appear since then
<sicelo> so no way to select 3G or 2G anymore ... but it should be a small thing. just need to make time to look into it
<Wizzup> wonder if that is a bug
<Wizzup> seems like it is
<Wizzup> uvos: so this is what the voicecall-manager pt plugin implements https://github.com/maemo-leste-upstream-forks/voicecall/blob/master/lib/src/abstractvoicecallhandler.h
<Wizzup> this is the interface I use atm from sphone https://github.com/maemo-leste-upstream-forks/voicecall/blob/master/lib/src/voicecallmanagerinterface.h but if we just load the plugin in sphone directly it won't be all that different I suppose
<Wizzup> uvos: we'd have to pull in some additional code if we would just use their interface as-is
<Wizzup> doesn't seem too hard though, if you think that makes more sense
<Wizzup> or we can hard-fork the code and just modify it to our needs and not care about their future fixes
<Wizzup> uvos: but I am finding more than yeah we might not want to just install sfos vc-mgr
<Wizzup> so we could write code for sphone to just load vc-mgr providers plugins directly, or copy the code to sphone, what do you think?
<Wizzup> copying directly would probably make more sense for us: easier to maintain / modify and such, and voicecall-mgr isn't normally packaged on debian anyway (I packaged it for us)
<Wizzup> uvos: in any case let me know what you think before I proceed to do useless work ;)
<Wizzup> freemangordon: ^^
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<freemangordon> Wizzup: maybe I am missing the details, but I think using upstream (and not forking) is more sane
<freemangordon> I doubt they will make drastic changes to the interface with out need
<freemangordon> packaging should not be a stopper
<freemangordon> not to say if we have that as a package, another UI will be more easy to create
<Wizzup> what do you mean with 'using upstream' specifically
<Wizzup> to use the tp .so from voicecall-manager but not the binary?
<Wizzup> as in, not run voicecall-manager, but use it's shared object directly
<Wizzup> the 'problem' with voicecall-manager is that it basically does what sphone does, and it actually also talks to mce and such (with wrong signatures sometimes), which we don't really want
<Wizzup> also I might have to make some changes to the tp plugin potentially
<Wizzup> but they (voicecall-manager) mostly abstracted the code quite well
<Wizzup> so the way I see it we have a few options...
<Wizzup> 1. talk to voicecall-manager over dbus from sphone (what I implemented currently)
<Wizzup> 2. implement stub voicecall-manager interface in sphone and load voicecall-manager .deb's .so directly (this is as upstream as it gets)
<Wizzup> 3. copy code from voicecall-manager to sphone and modify it a bit to suit our needs
<Wizzup> please note that I am 100% confident we will have to make a bunch of changes to the tp code regardless, it doesn't support jabber at all atm (protocol hardfilter on 'sip' and 'tel')
<Wizzup> there's also the anonymous caller bit being set
<Wizzup> I think (1) will get us quickly to TP calls, and I might press ahead just to make this work for my devices today/now
<Wizzup> I think (3) will be the easiest for us going forward, (2) would be more tricky since for every change we'd have to recompile the pkg
<Wizzup> freemangordon: ^
<sicelo> Wizzup: forgive the possibly silly question ... vcm is needed because it has tp integration/support already? or tp is still totally separate?
<Wizzup> yes, I was looking at it for TP
<Wizzup> but we can also just use their TP code in sphone and modify it, rather than run VCM
<Wizzup> and I have it working now with TP, but apart from a few minor small fixes there is an architectural thing where it replaces a lot of what sphone does for us
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<Wizzup> freemangordon: do you know if tracker has logs?
<Wizzup> I am also seeing poor pm and now my database is gone again, and tracker-extract was using a lot of cpu
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<freemangordon> Wizzup: I think logs are disabled by default
<freemangordon> isn't ther anything in syslog?
<freemangordon> Wizzup: re vcm - seems I am missing the details, however, wouldn't we want some cooperation with SF guys?
<freemangordon> like, they will support debian packaging and in return we will implement jabber or somesuch
<Wizzup> freemangordon: that doesn't solve the problems I mentioned
<Wizzup> also I found the sfos people to very unresponsive
<Wizzup> to be*
<freemangordon> I see
<Wizzup> freemangordon: I tried to give the details :D
<freemangordon> sure
<freemangordon> ok, I don;t understand what are the cons of option 1
<Wizzup> separate daemon that does all kinds of things like talk to mce, turn screen on, etc
<Wizzup> which sphone already does for us
<Wizzup> and the same downsides from (2) apply - regarding fixing things
<freemangordon> ok, but if I want to implement tphone someday, I will be able to focus on the UI only, no?
<Wizzup> what is tphone?
<freemangordon> s->t
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<Wizzup> well, not if you want the actual integration
<Wizzup> like, what about mce
<freemangordon> if vcm already talks mce, albeit some dialect, we shall fix/improve that
<Wizzup> why?
<Wizzup> what will do rtcom logging?
<freemangordon> because it we are to use another dialer someday, not sphone, we'll have to reimplement all this from scratch
<Wizzup> not really, the code will still be there
<freemangordon> where?
<freemangordon> in sphone?
<Wizzup> sure
<freemangordon> that does not make it reusable
<Wizzup> unless you plan to not use sphone I don't think having another daemon that does the same makes sense imo
<Wizzup> nevertheless, this is exactly what I did in pull request 4
<Wizzup> and I run into stupid things like I can't control the privacy/anonymous bit through the interface they offer
<Wizzup> also, for any video calls you'll also need some level of control
<Wizzup> as in, why would we try to 'fix' mce calls for sailfish (not maemo mce) when we have it working in sphone already, is what i mean
<Wizzup> we can still try to fix voicecall-manager but most of what it does, we do not need
<Wizzup> I'm pretty sure sfos just uses the ofono provider instead of the telepathy-ring one btw, but that is just a guess
<freemangordon> yes, I plan to not use sphone, unless uvos decides to implement/accept proper abook integration at some point and give up on his plans to remove rtcom support (iirc)
<Wizzup> we can make a hildon gtk2 ui and load it
<Wizzup> the ui is already modular
<freemangordon> it is not about the ui
<freemangordon> have to go to the store, brb
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<freemangordon> Wizzup: also, I am not sure sphone has any concept of video calls
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<Wizzup> freemangordon: I just want to get working calls with tp-ring and sip/jabber, what do you think is the best path forward, with not too much effort
<freemangordon> sphone
<freemangordon> but at the same time we shall try to make future replacement as easy as possible. not that 'easy' is applicable here, but I think you get the point
<Wizzup> sure
<Wizzup> it's quite funny to me just how hard it is to figure out what params can be set for a tp account
<Wizzup> I am trying to see if anon calls are just enabled by default in tp ring or something and I just can't figure it out
<Wizzup> freemangordon: in any case I still don't know how to proceed
<Wizzup> I am just going to try to debug the one problem with anon calls and then I can use this on my device to develop conversations with dsc for sms and such
<Wizzup> $ mdbus2 org.freedesktop.Telepathy.ConnectionManager.ring /org/freedesktop/Telepathy/Connection/ring/tel/ring org.freedesktop.DBus.Properties.GetAll org.freedesktop.Telepathy.Connection.Interface.Anonymity
<Wizzup> ({'AnonymityModes': <uint32 0>, 'SupportedAnonymityModes': <uint32 3>, 'AnonymityMandatory': <false>},)
<Wizzup> I suppose 0 is just do whatever is requested, or perhaps what the network wants
<Wizzup> perhaps my provider does anon by default is not specified
<Wizzup> what would explain why this is not a problem on my vm,but it is on my d4
<Wizzup> ok, now it works!
<Wizzup> mdbus2 org.freedesktop.Telepathy.ConnectionManager.ring /org/freedesktop/Telepathy/Connection/ring/tel/ring org.freedesktop.DBus.Properties.Set org.freedesktop.Telepathy.Connection.Interface.Anonymity AnonymityModes 2
<Wizzup> now sphone doesn't hide my caller id
<Wizzup> so voicecall-manager doesn't set this connection param
<Wizzup> that's a problem
<Wizzup> in any case, freemangordon, shall I continue with (1) until we decide a better wy?
<Wizzup> I personally think (3) is best fwiw, but it would take a bit more time, but provide us with flexibility
<Wizzup> we can always contribute fixes back to them if they are interested
<freemangordon> Wizzup: whatever fits you best
<Wizzup> there is also a way to set anonymity per channel
<Wizzup> freemangordon: well I might start asking you for help at some point with sip/jabber
<freemangordon> sure
<freemangordon> I am a bit overloaded with my RL job, but during the weekend I will have spare time
<Wizzup> sometimes trying to telepathy is quite frustrating :D
<freemangordon> heh
<ac_laptop> was there a dual-sim version of the n900 ?
<freemangordon> no
<ac_laptop> I stumbled upon this
<ac_laptop> https://www.ebay.com/itm/186171242846?hash=item2b58abb95e:g:LxQAAOSwykVlWTNy&amdata=enc%3AAQAIAAAA4K7a4ZYdsYv90p5Us9OmAlsrgOpyb48dOYfkDI4peeAgOICveHQaczaPIWZfONDj0JCNaMapNrK0Bre2Y3MjnEELh%2F%2FXUBip4UgdyzH37EsScjojkY9w4p94e3t7OkpCE5hi%2FOplO%2BurHM1uoobpzi%2FGOqpOzlTbvYKUm8YJgCmNgQj2akoaLs7ylWJSdP5jBsvAKrPPLW1W79jvlEdzzzPXFTzxc0rl0F21Vmt1Lh8SSJanLOgF0oIyl2h7x8zW0utcopa3371OVkqNiBD6r8tWea%2BhiB
<ac_laptop> H%2Bamudx8tHDRu6%7Ctkp%3ABk9SR6b-8fWCYw
<ac_laptop> sorry for the long link, here's the short one : https://www.ebay.com/itm/186171242846
<ac_laptop> anyway I guess it's modded hardware
<sicelo> ac_laptop: there were lots of fakes too, fyi :-p
<sicelo> as you've been told already, no real N900 was dual sim
<sicelo> poor people going to lose $50
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<uvos__> Wizzup: having a sphone module that can load vcm modules would be fine by me, as long as you dont have to "poision" sphone core, ie leak details of the vcm modue interface
<uvos__> freemangordon: what is "rtcom support" the only original rtcom piece used by sphone is rtcom-el, im not planning on dropping that, but eventually i want to add a module that logs allows you to its own sql database, but that dosent affect us here at all
<uvos__> i am planning on removeing the abook module
<uvos__> but that is because it dosent do anything that is usefull at this point
<uvos__> it opens a abook dialog to allow you to select a contact and phone number
<uvos__> but you might as well use that button in sphone to open osso-addressbook, wich allows you to do the same thing now, while being simply better than the abook dialog
<uvos__> Wizzup: if you want to do 3 thats fine to by me, but its more work, and in the end the qt code you then have inside of sphone will allways be pretty akward anyhow in a primary glib application
<uvos__> then again i want to replace the current gtk ui with qt, so this is sphones future anyhow, and is why the inter-module interfaces are all plain c
<uvos__> not "poision" sphone core ofc dosent mean not adding new functionality required, we just need to think of how to do generic interfaces that fit with all the other inter-module interfaces
<uvos__> im happy to do this part
<uvos__> that includes add video calls, atho i think this is really not a priority at all atm.
<uvos__> and will be hard to do in a system where i expect the provider of the video call and the ui to not leak toolkit details to eatch other, while avoiding copying things around.
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<Wizzup> uvos__: thanks, I am not sold on the loading vcm modules directly
<Wizzup> feels very hacky
<Wizzup> uvos__: to me (3) seems more future proof, we already have qt code in sphone now with telepathy qt and it works fine
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<uvos__> for 2 i was thinking having a shim.c file that we compile with the module sources to compile sphone native modules from vcm sources
<uvos__> not loading the vcm modules as is
<uvos__> but keeping the source files close enough to keep merging sfos patches in
<Wizzup> so sfos rarely touches this, like less than twice a year or so
<uvos__> Wizzup: sure @ qt code in sphone, its just akward as long as this is the only qt part, and the ui is still gtk, so mostly near-ish term
<uvos__> Wizzup: ok
<Wizzup> well I already did this in the current pull request (mapping qt to gtk)
<Wizzup> I mean their code doesn't even let you set anon calls or not
<uvos__> you mapped qt to sphones moudle interface
<uvos__> wich is not gtk
<Wizzup> so I bet they might not use it
<Wizzup> yeah, ok, sry :)
<uvos__> i mean we can evolve 2 into 3 slowly prety smoothly
<uvos__> so maybe we should do that
<uvos__> but whatever you like best
<Wizzup> I am not sure what the benefit of 2 would be if we don't plan to do it long term I think
<Wizzup> argh, now the modem of my VM resets when it goes online :)
<Wizzup> life is just full of struggles
<Wizzup> :D
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<Wizzup> freemangordon: I can't figure out how to make a channel using ensureChannel* or createChannel* and actually pass any properties on the channel
<Wizzup> telepathyprovider.cpp:136:135: error: use of deleted function ‘Tp::Client::ConnectionInterfaceAnonymityInterface::ConnectionInterfaceAnonymityInterface(Tp::Client::ConnectionInterfaceAnonymityInterface&&)’
<Wizzup> really
<Wizzup> I don't know how anyone ever just groks the TP APIs
<Wizzup> i'm just trying to set the anonymity bits using tp qt and it seems freaking impossible
<Wizzup> and a similar way for a channel