06:59 <freemangordon> arno11: please add 'Option "TripleBuffer" "true"' to /usr/share/X11/xorg.conf.d/99-omap.conf to see if it will make it smoother

07:50 <arno11> freemangordon: no diff with triple buffer

07:55 <arno11> the 'feeling' is exactly the same

13:15 <Wizzup> arno11: anything we need n900 serial for now btw?

13:16 <arno11> good question, don't remember if there is something else to look at.

13:17 <Wizzup> ok

13:18 <Wizzup> I'll be back in a few hours and will then look at seeing if I can update my ofono pr with the voicecall/telepathy work

13:18 <Wizzup> so then we should get VoIP as well soon :)

13:18 <arno11> oh really cool

13:22 <arno11> btw for voip, we need a solution for the daemon.conf to get audio (pkg or leste-config?)

13:37 <Wizzup> yes for n900

13:37 <Wizzup> and probably more tbh :)

13:37 <Wizzup> for tp sip I mean

13:37 <arno11> yes probably

18:30 <Wizzup> do others also see a bunch of this:

18:30 <Wizzup> error time=1701023567.586889 sender=org.freedesktop.DBus -> destination=:1.2 error_name=org.freedesktop.DBus.Error.AccessDenied reply_serial=210

18:30 <Wizzup> string "Rejected send message, 1 matched rules; type="error", sender=":1.2" (uid=0 pid=2233 comm="/usr/bin/mce --force-syslog ") interface="(unset)" member="(unset)" error name="org.freedesktop.DBus.Error.UnknownMethod" requested_reply="0" destination=":1.38" (uid=1000 pid=2864 comm="/usr/bin/hildon-home ")"

18:30 <Wizzup> in dbus-monitor --system

18:30 <Wizzup> og, this might actually be sfos doing something that our mce doesn't understand

18:46 <buZz> i let through all those replies to the ML about some binary switch

18:46 <buZz> and added the senders to 'accept'

18:50 <Wizzup> thanks

19:09 <Wizzup> huh, voicecall from sfos somehow doesn't pick up on telepathy-ring incoming calls, but outgoing works just fine

19:09 <Wizzup> pretty sure that wasn't a problem in beowulf

19:49 <Wizzup> freemangordon: uvos: so I have sphone with some hacks now able to dial out using telepathy-ring, telepathy-gabble and telepathy-rakia, but there are still problems to solve:

19:49 <Wizzup> this is with sfos voice call in between

19:49 <Wizzup> 1. incoming calls are not seen by sfos voicecall manager when using telepathy-sip

19:50 <Wizzup> 2. calls don't actually work with xmpp yet, there is a state mismatch (this is not weird, sfos voicecall doesn't actually support it, but it might not be too hard)

19:51 <Wizzup> 3. incoming sip I didn't test yet mostly due to some weirdness, could use some help testing this

19:51 <Wizzup> there are probably a bunch of gstreamer things to figure out, but the main issue right now is that somehow incoming calls from telepathy-ring are not seen, and I'm sure that used to work, so something changed in tp-ring perhaps, or somewhere else

19:52 <Wizzup> but it's kind of cool to be able to type a sip or xmpp address in sphone, select the right backend, and have it actually dial out

19:52 <sicelo> n900 & serial - still needed for off mode

19:53 <Wizzup> sicelo: if you let me know what to do/run I'll do it

19:53 <sicelo> i was relying more on you for that :-p

19:54 <sicelo> i mean, i was going to pick up from the work you've done in the past

19:57 <arno11> (for sip and xmpp) really cool

20:12 <Wizzup> sicelo: I don't remember actually getting a trace when off mode resets

20:12 <Wizzup> arno11: cool but not for enough alnog yet

20:12 <Wizzup> along

20:13 <arno11> good start anyway

20:33 <Wizzup> sicelo: that was the problem: keeping serial on = no off mode, detaching serial = no trace

20:33 <Wizzup> sicelo: IIRC

20:55 <bencoh> sounds like a jtag would help here

20:57 <bencoh> except that I can't find any jtag pin in the n900 schematics .... :/

21:03 <Wizzup> it might just be that I don't have the stun/nat etc working well and this is why all sip/xmpp fails

21:04 <Wizzup> I use the default user networking in qemu, so that doesn't help I think

21:49 <Wizzup> it looks like ring creates a channel as StreamedMedia, and voicecall doesn't explicitly request this type, but I am not sure

22:09 <Wizzup> yup, dunno for now, giving up for today

22:09 <Wizzup> I was pretty sure this was working before

22:10 <Wizzup> there is a chance the difference is d4 vs my vm with usb modem

22:10 <bencoh> I dunno how qemu usernet handles sip/rtp and nat traversal

22:11 <bencoh> I doubt it has any protocol-specific conntrack

22:11 <bencoh> (which is usually required for sip/rtp)

22:12 <Wizzup> when I mean 'the difference' I mean that perhaps somehow the usb quectel modem call in tp ring is different from how it works on the d4

22:12 <Wizzup> seems unlikely

22:12 <Wizzup> qemu usernet indeed isn't great and it is probably the cause of my some of my sip/xmpp troubles

22:13 <bencoh> wait, so does your vm see the quectel modem as a usb device, or not?

22:13 <Wizzup> but nevermind sip/xmpp,if we can't get incoming tp ring call announced for some reason

22:13 <Wizzup> bencoh: yes

22:13 <bencoh> I see

22:13 <Wizzup> handleChannels() is not called on voicecall-qt5 for some reason for the incoming channel

22:13 <Wizzup> and I'm pretty sure it should

22:14 <Wizzup> Looking at https://github.com/maemo-leste/sphone/pull/4 I wrote in the pull request

22:14 <Wizzup> - telepathy-ring calls just work - call hold is not tested

22:15 <Wizzup> looks like I was testing in chimaera

22:19 <bencoh> how do you know it was chimaera?

22:20 <Wizzup> by the date, and the fact that pgkweb only shows voicecall-qt5 for chimaera

22:20 <Wizzup> https://maedevu.maemo.org/pkgweb/search?q=voicecall-qt5

22:20 <bencoh> ah

22:20 <bencoh> well

22:20 <Wizzup> well, I did all of the development on my d4

22:20 <Wizzup> I guess there really is a chance that ofono is just slightly different there somehow

22:21 <Wizzup> like maybe there's some weird VoLTE stuff going on somehow, idk

22:21 <bencoh> silly question, but why do we need voiceall-manager on top of telepathy-ring and ofono?

22:22 <Wizzup> because it already solves the problem we're trying to solve

22:22 <bencoh> oh, hmm, volte might require sw support, depending on the modem implem

22:22 <Wizzup> we can also copy-paste their code (4500 loc) and try to integrate it into sphone somehow

22:22 <bencoh> but assuming you're using the same modem as the one in the pinephone, I kinda understood that it should work out of the box(I think?)

22:23 <Wizzup> volte is probably a red herring

22:24 <Wizzup> there's just something weird going on where mission control decides that the voicecall code should not get to handle this channel that tp-ring creates for an incoming call

22:24 <Wizzup> and tbh, the channel being StreamedMedia is a little weird

22:25 <Wizzup> although maybe that's fine for all calls

22:26 <bencoh> maybe you could compare with a working sailfishos

22:27 <Wizzup> maybe if I'm stuck for a long time

22:49 <ac_laptop> hello people

22:50 <ac_laptop> I'm once again in one of those situations where I plug a battery that is 100% full, boot under fremanlte, it boots fine and shows a battery at 100%, boot under maemo, it goes to the desktop screen and then shuts down/crashes. What should I do ?

22:52 <ac_laptop> 1) plug the device when off 2) plug the device under fremantle 3) plug the device under leste, 4) try to get to logs

22:57 <Wizzup> bencoh: ha, yep

22:57 <Wizzup> it 'just works' on the d4

22:57 <Wizzup> no idea wy

22:58 <Wizzup> sicelo: uvos: did we fix the anonymous caller bit in ofono on the d4?

22:59 <Wizzup> the outgoing calls on d4 still say 'anonymous caller'

22:59 <Wizzup> (coul be tpqt related)

23:00 <Wizzup> https://github.com/maemo-leste-upstream-forks/ofono/commit/e0af50a500c75369c31e211bfe8c82d6c164a4e8 looks like we did

23:04 <uvos__> Wizzup: yes we did, you need up to date ofono and sphone

23:04 <Wizzup> I think I do

23:04 <Wizzup> I was doing tp testing

23:05 <uvos__> bencoh: "silly question, but why do we need voiceall-manager on top of telepathy-ring and ofono?" yeah i dont like this either, ofono, ring, v-m and the sphone module are just an insane number of mostly redundant abstraction layers

23:06 <Wizzup> integrating the same code would still give the same bugs, so those need to be solved regardless

23:07 <Wizzup> so for some reason, tp ring + voicecall-manager on vm: - outgoing calls works great, no anonymous bit. -incoming calls aren't even seen

23:07 <Wizzup> on d4: -outgoing calls work but have privacy bit set somehow - incoming calls are seen and work great

23:08 <uvos__> sphone-ofono path works fine i presume

23:08 <uvos__> ?

23:09 <Wizzup> will try

23:10 <Wizzup> yes then it's not hidden

23:11 <Wizzup> uvos__: bencoh: fwiw if someone does the work to take the code from voicecall manager and make it into a sphone module I'll be happy to drop what I've been doing

23:11 <Wizzup> but until then I'm just going to stay the course and worry about this later

23:18 <uvos__> Wizzup: ok v-m doing somethong wierd then

23:19 <Wizzup> uvos__: it's the same modem as pinephone

23:19 <Wizzup> fwiw

23:19 <uvos__> v-m = voicecall manager

23:19 <uvos__> not virtual machine

23:19 <Wizzup> oh

23:19 <Wizzup> righ

23:20 <Wizzup> yeah will check

23:20 <uvos__> well my message was confusing

23:20 <Wizzup> but I'm done with this for today :D

23:20 <Wizzup> still cool that xmpp/sip works as much as it does already

23:21 <Wizzup> like I could type in xmpp or sip user and call them (calls failed in gst or some state change, but hey)

23:21 <Wizzup> (type them in in sphone)

23:40 <Wizzup> tmlind: uvos__: got a reset on 6.1.48: https://paste.villavu.com/raw/gYEEtB2LZEpLpfp0W3y2LjAcsWwTa3Jqagx0porb/