savoirfairelinux issueshttps://git.jami.net/groups/savoirfairelinux/-/issues2020-10-23T01:55:45Zhttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/821Dark theme issue2020-10-23T01:55:45ZSébastien BlinDark theme issue![Capture+_2020-10-22-18-58-27](/uploads/62608581a3f3d7bf1a500bdf1ceae59e/Capture+_2020-10-22-18-58-27.png)
Decline/Block are not readable
(also Invitation received is shown a lot)![Capture+_2020-10-22-18-58-27](/uploads/62608581a3f3d7bf1a500bdf1ceae59e/Capture+_2020-10-22-18-58-27.png)
Decline/Block are not readable
(also Invitation received is shown a lot)https://git.jami.net/savoirfairelinux/jami-project/-/issues/1085Feature request folder categorisation of chats2021-04-24T09:05:18ZFeature request folder categorisation of chatsI would like to organize my chats to folders like family, friends, work...I would like to organize my chats to folders like family, friends, work...https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/815Feature request: reduce the photo size before sending it2020-10-17T18:59:24ZDimitrios ApostolouFeature request: reduce the photo size before sending itWith recent phones, the photo can be as large as 5 to 10 MB in size. This causes the other side needing to approve the photo, which usually results to missed photos.
It would be nice to auto-reduce the photo size or quality before sending.With recent phones, the photo can be as large as 5 to 10 MB in size. This causes the other side needing to approve the photo, which usually results to missed photos.
It would be nice to auto-reduce the photo size or quality before sending.https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/814Feature request: enable loudspeaker while call is being routed2023-02-20T20:14:40ZDimitrios ApostolouFeature request: enable loudspeaker while call is being routedIn Google Hangouts, I can route audio output to the loudspeaker and hear the ringing tone until the other person picks up.
In Jami, I have to wait on the device for maybe a minute or two (it takes long for the call to get through) until...In Google Hangouts, I can route audio output to the loudspeaker and hear the ringing tone until the other person picks up.
In Jami, I have to wait on the device for maybe a minute or two (it takes long for the call to get through) until the other side picks up, and then I am able to change to the loudspeaker.
It would be nice to be able to set this in advance.https://git.jami.net/savoirfairelinux/jami-jams/-/issues/59Search directory endpoint should return 502 error2020-10-16T18:26:28ZLarbi GharibSearch directory endpoint should return 502 errorSteps to reproduce the issue:
Contact search directory endpoint with and expired token.
Expected behaviour:
Error 401 returned with json
Actual behaviour:
Error 502 return with HTMl pageSteps to reproduce the issue:
Contact search directory endpoint with and expired token.
Expected behaviour:
Error 401 returned with json
Actual behaviour:
Error 502 return with HTMl pageLarbi GharibLarbi Gharibhttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/811Feature request: call recording on Android client2020-10-15T12:01:06ZFeature request: call recording on Android clientCall recording should be available on phonesCall recording should be available on phoneshttps://git.jami.net/savoirfairelinux/jami-packaging/-/issues/81Enhance dependency tracking2020-10-02T16:04:37ZPete GossnerEnhance dependency trackingsee: https://review.jami.net/c/ring-project/+/12428
> run_dependencies() should be changed instead.
FWIW I agree.
- The current package listings
1/ requires potential developers to run older and static releases. (debian stable)
2/ s...see: https://review.jami.net/c/ring-project/+/12428
> run_dependencies() should be changed instead.
FWIW I agree.
- The current package listings
1/ requires potential developers to run older and static releases. (debian stable)
2/ should exploit any system apt tool kit more particularly.
I would also be willing to attempt this for at least the apt based distros by extending the existing python tool to more fully exploit apt-get and apt-cache (or dpkg) or even require apt-file.
! I would work on a local branch !https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/805jams - incorrect name displayed if user has no displayname2020-09-25T15:47:53ZGuillaume Hellerjams - incorrect name displayed if user has no displaynameConnect with a user who has no first name and no last name
--> Jami returns "Compte Jami 3"
--> if no displayname is available, we should fallback and display the usernameConnect with a user who has no first name and no last name
--> Jami returns "Compte Jami 3"
--> if no displayname is available, we should fallback and display the usernameAdrien BéraudAdrien Béraudhttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/803Some images are not displayed2020-09-18T14:51:34ZCyrille BéraudSome images are not displayedsend a png file to a Jami/android - the image is received but not diplayedsend a png file to a Jami/android - the image is received but not diplayedAdrien BéraudPierre DucheminAdrien Béraudhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1057No early media after SIP 1832020-10-02T21:14:46ZPaweł BogusławskiNo early media after SIP 183Hello,
When using
* jami_20200829.1.d8786b1~dfsg1-1_amd64.deb
* jami-daemon_20200829.1.d8786b1~dfsg1-1_amd64.deb
from
https://dl.jami.net/nightly/debian_10/pool/main/j/jami/
in Debian 10, Jami does not play ringback when calling ext...Hello,
When using
* jami_20200829.1.d8786b1~dfsg1-1_amd64.deb
* jami-daemon_20200829.1.d8786b1~dfsg1-1_amd64.deb
from
https://dl.jami.net/nightly/debian_10/pool/main/j/jami/
in Debian 10, Jami does not play ringback when calling external number (which uses SIP 183 message when waiting for call to be picked up and early media/SDP); when call is picked up - voice is played correctly; dring debug log of such call:
```
[1600086622.792|15229|manager.cpp :581 ] ----- Switch current call id to '2159493006188118' -----
[1600086622.792|15230|sipcall.cpp :963 ] [call:2159493006188118] fill SDP with ICE transport 0x5633ef823740
[1600086622.792|15230|sipaccount.cpp :375 ] contact header: "test" <sip:login@192.168.1.100:5060> / "test" <sip:login@sip.mydomain.loc> -> <sip:111222333@sip.mydomain.loc>
[1600086622.792|15230|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 1 (CALLING): cause=0, tsx@0x7fd6dc048e88 status 0 (Default status message)
[1600086622.793|15230|call.cpp :259 ] [call:2159493006188118] state change 0/1, cnx 0/2, code 0
[1600086622.793|15230|call.cpp :286 ] [call:2159493006188118] emit client call state change CONNECTING, code 0
[1600086624.793|15230|call.cpp :112 ] Call 2159493006188118 is still connecting after timeout, sending fallback request
[1600086625.482|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 3 (EARLY): cause=0, tsx@0x7fd70800d278 status 183 (Session Progress)
[1600086625.482|15233|sipvoiplink.cpp :1009 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 media update: status 0
[1600086625.482|15233|sdp.cpp :383 ] Local active SDP Session:
v=0
o=ibnb003 3809075422 0 IN IP4 192.168.1.100
s=Jami Daemon
c=IN IP4 192.168.1.100
t=0 0
a=ice-ufrag:1d26a479
a=ice-pwd:33f8ac4760b4c6501dd61dfa
m=audio 41884 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:41885 IN IP4 192.168.1.100
a=sendrecv
a=candidate:H44e08c2a 1 UDP 2130706431 [...] 42537 typ host
a=candidate:Ha030271 1 UDP 2130706431 192.168.1.100 39418 typ host
a=candidate:Hc0a87b64 1 UDP 2130706175 [...] 39418 typ host
a=candidate:Hc0a87a01 1 UDP 2130705919 [...] 39418 typ host
a=candidate:H44e08c2a 2 UDP 2130706430 [...] 42180 typ host
a=candidate:Ha030271 2 UDP 2130706430 [...] 35281 typ host
a=candidate:Hc0a87b64 2 UDP 2130706174 [...] 35281 typ host
a=candidate:Hc0a87a01 2 UDP 2130705918 [...] 35281 typ host
m=video 0 RTP/AVP 96 97 98 99
[1600086625.482|15233|sdp.cpp :383 ] Remote active SDP Session:
v=0
o=ib 1858813877 1858813877 IN IP4 192.168.100.2
s=IB VoIP Server
c=IN IP4 192.168.100.2
t=0 0
m=audio 23562 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=maxptime:60
a=sendrecv
m=video 0 RTP/AVP 96 97 98 99
[1600086625.482|15233|sipcall.cpp :1230 ] [call:2159493006188118] medias changed
[1600086625.482|15230|sipcall.cpp :1242 ] [call:2159493006188118] no remote ICE for medias
[1600086625.482|15230|sipcall.cpp :1182 ] [call:2159493006188118] stopping all medias
[1600086625.482|15230|sipcall.cpp :1021 ] [call:2159493006188118] startAllMedia()
[1600086625.482|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41884
[1600086625.482|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41885
[1600086625.482|15230|socket_pair.cpp :349 ] SocketPair: local{41884,41885} / 192.168.100.2{23562,23563}
[1600086625.482|15230|audio_input.cpp :53 ] Creating audio input with id: 2159493006188118
[1600086625.483|15230|audio_input.cpp :270 ] Switching audio source to match 'camera://04f2b3ec200901010001'
[1600086625.483|15230|audio_sender.cpp :72 ] audioEncoder_->openOutput rtp://192.168.100.2:23562
[1600086625.483|15230|media_encoder.cpp :268 ] Not using hardware encoding for speex
[1600086625.483|15230|media_encoder.cpp :574 ] [libspeex] Using 4 threads
[1600086625.483|15230|media_encoder.cpp :621 ] [libspeex] Frame size 160
[1600086625.483|15230|sipcall.cpp :1065 ] [call:2159493006188118] [SDP:slot#1] Missing local codec
[1600086625.483|16375|media_decoder.cpp :134 ] Trying to open device dummyFilename with format sdp, pixel format , size 0x0, rate 0.000000
[1600086625.483|16375|media_decoder.cpp :146 ] Using format sdp
[1600086625.503|16374|audio_input.cpp :76 ] Switching audio input to '04f2b3ec200901010001'
[1600086628.306|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 4 (CONNECTING): cause=0, tsx@0x7fd70800d278 status 200 (OK)
[1600086628.306|15233|sipvoiplink.cpp :1009 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 media update: status 0
[1600086628.306|15233|sdp.cpp :383 ] Local active SDP Session:
v=0
o=ibnb003 3809075422 1 IN IP4 192.168.1.100
s=Jami Daemon
c=IN IP4 192.168.1.100
t=0 0
a=ice-ufrag:1d26a479
a=ice-pwd:33f8ac4760b4c6501dd61dfa
m=audio 41884 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:41885 IN IP4 192.168.1.100
a=sendrecv
a=candidate:H44e08c2a 1 UDP 2130706431 [...] 42537 typ host
a=candidate:Ha030271 1 UDP 2130706431 [...] 39418 typ host
a=candidate:Hc0a87b64 1 UDP 2130706175 [...] 39418 typ host
a=candidate:Hc0a87a01 1 UDP 2130705919 [...] 39418 typ host
a=candidate:H44e08c2a 2 UDP 2130706430 [...] 42180 typ host
a=candidate:Ha030271 2 UDP 2130706430 [...] 35281 typ host
a=candidate:Hc0a87b64 2 UDP 2130706174 [...] 35281 typ host
a=candidate:Hc0a87a01 2 UDP 2130705918 [...] 35281 typ host
m=video 0 RTP/AVP 96 97 98 99
[1600086628.306|15233|sdp.cpp :383 ] Remote active SDP Session:
v=0
o=ib 1858813877 1858813877 IN IP4 192.168.100.2
s=IB VoIP Server
c=IN IP4 192.168.100.2
t=0 0
m=audio 23562 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=maxptime:60
a=sendrecv
m=video 0 RTP/AVP 96 97 98 99
[1600086628.306|15233|sipcall.cpp :1230 ] [call:2159493006188118] medias changed
[1600086628.306|15230|sipcall.cpp :1242 ] [call:2159493006188118] no remote ICE for medias
[1600086628.306|15230|sipcall.cpp :1182 ] [call:2159493006188118] stopping all medias
[1600086628.306|15233|sipvoiplink.cpp :861 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 5 (CONFIRMED): cause=0 (TX_MSG)
[1600086628.306|15233|sipcall.cpp :910 ] [call:2159493006188118] onAnswered()
[sdp @ 0x7fd6bc0206c0] Could not find codec parameters for stream 0 (Audio: speex (libspeex), 8000 Hz, mono): unspecified sample format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[1600086628.387|16375|media_decoder.cpp :481 ] Decoding audio using libspeex Speex (libspeex)
[1600086628.387|16375|media_decoder.cpp :491 ] Not using hardware decoding for speex
[1600086628.387|15230|sipcall.cpp :1021 ] [call:2159493006188118] startAllMedia()
[1600086628.387|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41884
[1600086628.387|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41885
[1600086628.387|15230|socket_pair.cpp :349 ] SocketPair: local{41884,41885} / 192.168.100.2{23562,23563}
[1600086628.387|15230|audio_input.cpp :270 ] Switching audio source to match 'camera://04f2b3ec200901010001'
[1600086628.387|15230|audio_sender.cpp :72 ] audioEncoder_->openOutput rtp://192.168.100.2:23562
[1600086628.387|15230|media_encoder.cpp :268 ] Not using hardware encoding for speex
[1600086628.387|15230|media_encoder.cpp :574 ] [libspeex] Using 4 threads
[1600086628.387|15230|media_encoder.cpp :621 ] [libspeex] Frame size 160
[1600086628.387|15230|sipcall.cpp :1065 ] [call:2159493006188118] [SDP:slot#1] Missing local codec
[1600086628.387|16376|media_decoder.cpp :134 ] Trying to open device dummyFilename with format sdp, pixel format , size 0x0, rate 0.000000
[1600086628.387|15230|call.cpp :259 ] [call:2159493006188118] state change 1/1, cnx 2/4, code 0
[1600086628.387|15230|call.cpp :286 ] [call:2159493006188118] emit client call state change CURRENT, code 0
[1600086628.387|16376|media_decoder.cpp :146 ] Using format sdp
[1600086628.387|15230|manager.cpp :2007 ] [call:2159493006188118] Peer answered
[1600086628.387|15230|manager.cpp :1631 ] Add audio to call 2159493006188118
[1600086628.387|15230|manager.cpp :1645 ] [call:2159493006188118] Attach audio
[1600086628.387|15230|ringbufferpool.cpp:175 ] Bind call 2159493006188118 to call audiolayer_id
[1600086628.387|15230|ringbufferpool.cpp:156 ] Bind rbuf '2159493006188118' to callid 'audiolayer_id'
[1600086628.387|15230|ringbufferpool.cpp:156 ] Bind rbuf 'audiolayer_id' to callid '2159493006188118'
[1600086628.387|15230|audiolayer.cpp :130 ] Hardware audio format available : {s16, 2 channels, 44100Hz} 0
[1600086628.387|15230|audiostream.cpp :53 ] Playback: trying to create stream with device alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.387|15230|audiostream.cpp :53 ] Ringtone: trying to create stream with device (44100Hz, 2 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.387|15230|audiostream.cpp :53 ] Capture: trying to create stream with device alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.403|16374|audio_input.cpp :76 ] Switching audio input to '04f2b3ec200901010001'
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.pci-0000_00_1b.0.analog-stereo
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600086629.465|15245|pulselayer.cpp :371 ] All streams ready, starting !
[1600086629.588|16376|media_decoder.cpp :481 ] Decoding audio using libspeex Speex (libspeex)
[1600086629.588|16376|media_decoder.cpp :491 ] Not using hardware decoding for speex
[1600086634.112|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 6 (DISCONNCTD): cause=200, tsx@0x7fd70800d278 status 200 (OK)
[1600086634.112|15230|manager.cpp :2040 ] [call:2159493006188118] Peer hungup
```
Problem does not occur on same Debian system and SIP account when using Twinkle SIP client.
Problem does not occur in Jami when local VoiP number is called (which uses SIP 180 message when waiting for call, without early media/SDP); dring debug log of such /not answered/ call when ringback was played correctly:
```
[1600085689.195|15229|manager.cpp :581 ] ----- Switch current call id to '5190575654160253' -----
[1600085689.195|15230|sipcall.cpp :963 ] [call:5190575654160253] fill SDP with ICE transport 0x5633ef824e50
[1600085689.195|15230|sdp.cpp :718 ] addIceCandidates failed: cannot access media#1 (may be deactivated)
[1600085689.196|15230|sdp.cpp :718 ] addIceCandidates failed: cannot access media#1 (may be deactivated)
[1600085689.196|15230|sipaccount.cpp :375 ] contact header: "test" <sip:login@129.168.1.100:5060> / "IB" <sip:login@sip.mydomain.loc> -> <sip:login2@sip.mydomain.loc>
[1600085689.196|15230|sipvoiplink.cpp :853 ] [call:5190575654160253] INVITE@0x7fd6dc0460b8 state changed to 1 (CALLING): cause=0, tsx@0x7fd6dc0286b8 status 0 (Default status message)
[1600085689.196|15230|call.cpp :259 ] [call:5190575654160253] state change 0/1, cnx 0/2, code 0
[1600085689.196|15230|call.cpp :286 ] [call:5190575654160253] emit client call state change CONNECTING, code 0
[1600085689.374|15233|sipvoiplink.cpp :853 ] [call:5190575654160253] INVITE@0x7fd6dc0460b8 state changed to 3 (EARLY): cause=0, tsx@0x7fd70800d278 status 180 (Ringing)
[1600085689.374|15233|call.cpp :259 ] [call:5190575654160253] state change 1/1, cnx 2/3, code 0
[1600085689.374|15233|call.cpp :286 ] [call:5190575654160253] emit client call state change RINGING, code 0
[1600085689.374|15230|manager.cpp :2029 ] [call:5190575654160253] Peer ringing
[1600085689.374|15230|audiolayer.cpp :130 ] Hardware audio format available : {s16, 2 channels, 44100Hz} 0
[1600085689.374|15230|audiostream.cpp :53 ] Playback: trying to create stream with device alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085689.374|15230|audiostream.cpp :53 ] Ringtone: trying to create stream with device (44100Hz, 2 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085689.374|15230|audiostream.cpp :53 ] Capture: trying to create stream with device alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.pci-0000_00_1b.0.analog-stereo
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600085690.454|15245|pulselayer.cpp :371 ] All streams ready, starting !
[1600085697.904|15229|manager.cpp :1665 ] [call:5190575654160253] Remove local audio
[1600085697.904|15229|ringbufferpool.cpp:242 ] Unbind call 5190575654160253 from all bound calls
[1600085697.904|15229|sipcall.cpp :368 ] [call:5190575654160253] Terminate SIP session
```
It seems that Jami has problem playing early media from SDP after 183 SIP message (opening sound device problem maybe?) and has not such problem when self playing ringback tone after 180 SIP message.
Sound configuration is pulseaudio, rindtone device = default (uses standard speakers), output and input devices = Jabra PRO 930 Mono and works ok in other functions (i.e. incomming ringing on standard speakers but ringback tones and speech on Jabra).
Regards,
Pawełhttps://git.jami.net/savoirfairelinux/jami-jams/-/issues/55Read user agent when client connects to JAMS2020-09-08T15:02:12ZWilliam EnrightRead user agent when client connects to JAMSWhen a Jami client attempts to connect to JAMS, we should read the user agent in the device registration request and send back the appropriate configuration.
NOTE: This is for next release.When a Jami client attempts to connect to JAMS, we should read the user agent in the device registration request and send back the appropriate configuration.
NOTE: This is for next release.https://git.jami.net/savoirfairelinux/jami-daemon/-/issues/301No relay (Turn) candidate in SDP with IOS when being connected in LTE/4G2021-02-17T16:19:15ZCyrille BéraudNo relay (Turn) candidate in SDP with IOS when being connected in LTE/4GSee attached file for log (with an Android comparison)[sdpios-andoid.log](/uploads/891e408417fd3059abed50184563f396/sdpios-andoid.log)
To reproduce:
Be connected on LTE/4G, make a call.
btw, why all the addresses are twice as candidate?See attached file for log (with an Android comparison)[sdpios-andoid.log](/uploads/891e408417fd3059abed50184563f396/sdpios-andoid.log)
To reproduce:
Be connected on LTE/4G, make a call.
btw, why all the addresses are twice as candidate?BacklogMohamed ChibaniKateryna KostiukSébastien BlinMohamed Chibanihttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/794Feature Request: Visible Error Message2020-09-03T20:01:13ZMarinus SavoritiasFeature Request: Visible Error MessageWhen Jami connectivity fails or the file transfer fails. There is no clear message that it has indeed failed.
The user is left wondering if something is wrong with his client or the client he is trying to send to.
It would be beneficial ...When Jami connectivity fails or the file transfer fails. There is no clear message that it has indeed failed.
The user is left wondering if something is wrong with his client or the client he is trying to send to.
It would be beneficial to have some kind of error message saying that jami has crashed.
So that the user knows what to do.https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/791Screenshots are outdated in Play Store.2020-09-02T11:05:01ZMarinus SavoritiasScreenshots are outdated in Play Store.The Screenshots on Play Store showing the UI are outdated.The Screenshots on Play Store showing the UI are outdated.https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/787Feature Request: Contact Folders2020-09-03T19:55:21ZMarinus SavoritiasFeature Request: Contact FoldersWhen you add a lot of contacts in Jami things could get messy pretty quickly with a lot of people.
I know there is the search functionality but it would be also nice to implement contact folders like Telegram.
WDYT?When you add a lot of contacts in Jami things could get messy pretty quickly with a lot of people.
I know there is the search functionality but it would be also nice to implement contact folders like Telegram.
WDYT?https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/786Video Call button still works even though I disabled video from settings.2020-09-02T10:55:51ZMarinus SavoritiasVideo Call button still works even though I disabled video from settings.I disabled the video in media under settings. After that the video call button was still present and I still could perform a call. I got a video call screen with both windows black.
There shouldn't be a video call button if you have dis...I disabled the video in media under settings. After that the video call button was still present and I still could perform a call. I got a video call screen with both windows black.
There shouldn't be a video call button if you have disabled video.
**This issue is observed with:**
* Ring version: 20200810
* Device model: Fairphone 3
* Android version: 9.0 /e/OS with Microg
* What build you are using: F-Droidhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1042Rendez-vous with password protection2021-12-01T15:59:22Zring userRendez-vous with password protectionIt would be great to have a password protection for Rendez-vous.
There is already the "Allow calls from untrusted peers" option, but in my opinion it is not a good alternative for a password to prevent Jami-bombing because with "Allow c...It would be great to have a password protection for Rendez-vous.
There is already the "Allow calls from untrusted peers" option, but in my opinion it is not a good alternative for a password to prevent Jami-bombing because with "Allow calls from untrusted peers", you need to collect all allowed contacts before they can join. With a password, it is only necessary to give it to all persons you want to join and even if you didn't have them in your contacts, they can join, so it is much less effort.
See discussion in issue https://git.jami.net/savoirfairelinux/ring-daemon/issues/281.https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/782TV: bad preview ratio2023-06-01T17:33:43ZPierre DucheminTV: bad preview ratio- Jami version: 20200810-01
- Device model: Mecool KM9Pro Deluxe
- Android version: 10
- What build you are using: c1158499d97b9f24843617a44402c148e6bff08e + daemon: 27668c3a564c22e8a1971d7f9467f0a58b615990
## Steps to reproduce...- Jami version: 20200810-01
- Device model: Mecool KM9Pro Deluxe
- Android version: 10
- What build you are using: c1158499d97b9f24843617a44402c148e6bff08e + daemon: 27668c3a564c22e8a1971d7f9467f0a58b615990
## Steps to reproduce
- Can you reproduce the bug: occasionally
- Steps:
1. call Android TV device from jami-gnome (2020-08-10)
- Actual result: in some cases, the preview is malformed: bad orientation and inverted width and height. After a while the preview freeze.
![device-2020-08-13-164350](/uploads/264312e69249a07f37c90eb408690f8a/device-2020-08-13-164350.png)
- Expected result: the preview is showing as usual.
![device-2020-08-13-165925](/uploads/b231ca1a8a11418e258343123f1d2069/device-2020-08-13-165925.png)
## Additional information
[badpreviewratiotv.log](/uploads/09ece7be49e9613f4cba00028f755c2b/badpreviewratiotv.log)https://git.jami.net/savoirfairelinux/jami-project/-/issues/1039Add forbidden/black list on SIP call2020-08-17T03:08:00ZKmAdd forbidden/black list on SIP callHello
In some case we can get some unwanted contact. (as SIP bot).
Could be interesting to set a list to forbid these accounts.
In my use case, I get a call more or less each open day hours from same bot. Forbid it could be nice :)
Th...Hello
In some case we can get some unwanted contact. (as SIP bot).
Could be interesting to set a list to forbid these accounts.
In my use case, I get a call more or less each open day hours from same bot. Forbid it could be nice :)
Thankshttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/778TV: call hanging up after 2 seconds2023-06-01T17:34:45ZPierre DucheminTV: call hanging up after 2 seconds- Jami version: 20200722-01
- Device model: Mecool KM9Pro Deluxe
- Android version: 10
- What build you are using: client: 175a9b425068d4a1d18c58a29e8233a38b9353f0 + daemon: 4357af81409d209f0208f9b0b59059cd244d7b54
## Steps to r...- Jami version: 20200722-01
- Device model: Mecool KM9Pro Deluxe
- Android version: 10
- What build you are using: client: 175a9b425068d4a1d18c58a29e8233a38b9353f0 + daemon: 4357af81409d209f0208f9b0b59059cd244d7b54
## Steps to reproduce
- Can you reproduce the bug: at will
- Steps:
0. Enable auto-answer on Android TV device
1. kill/restart jami-gnome or jami android
2. call Android TV device from jami-gnome (2020-07-24) or jami android (20200715-01)
- Actual result: after exactly 2 seconds, the Android TV device hang up.
- Expected result: the call starts normally
## Additional information
- Everything works fine if the Android TV device calls jami-gnome or jami android
- race condition?
- Call log:
[callhangingup.log](/uploads/e72e1c791109a4d7cf339c1d66690925/callhangingup.log)