savoirfairelinux issueshttps://git.jami.net/groups/savoirfairelinux/-/issues2022-11-23T14:59:36Zhttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/786SIP connections are not gracefully terminated2022-11-23T14:59:36ZMaxim CournoyerSIP connections are not gracefully terminatedWhen using a SIP account, I noticed that upon either disabling the account in the settings or quitting the client, the server hasn't received the BYE and assumes the client is still connected (that's at least the case for voip.ms).
linp...When using a SIP account, I noticed that upon either disabling the account in the settings or quitting the client, the server hasn't received the BYE and assumes the client is still connected (that's at least the case for voip.ms).
linphone doesn't exhibit this problem.Antoine NoreauAntoine Noreauhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/720Support DTMF2023-06-16T18:56:18ZSébastien BlinSupport DTMFTo be definedTo be definedBackloghttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/694ffmpeg: add G711, G729 codecs (SIP)2022-07-06T00:18:18Zovariffmpeg: add G711, G729 codecs (SIP)Please add the following codecs to Jami:
* [G.711a](https://en.wikipedia.org/wiki/G.711)
* [G.711u](https://en.wikipedia.org/wiki/G.711)
* [G.729a](https://en.wikipedia.org/wiki/G.729)
What do you think?
Thank you
> MyNetFone uses th...Please add the following codecs to Jami:
* [G.711a](https://en.wikipedia.org/wiki/G.711)
* [G.711u](https://en.wikipedia.org/wiki/G.711)
* [G.729a](https://en.wikipedia.org/wiki/G.729)
What do you think?
Thank you
> MyNetFone uses the **G.711a** codec, it is the standard codec used in Australia and Europe. The **G.711u** codec is used within the US, and may be present in PBXs or UC platforms from US vendors. The **G.729a** codec is often used for conference calling services.<br>
> https://www.mynetfone.com.au/support/faq/question/477
> The patents for G.711, released in 1972, have expired, so it may be used without the need for a licence
> https://en.wikipedia.org/wiki/G.711#Licensing
> As of January 1, 2017, the patent terms of most licensed patents under the G.729 Consortium have expired, the remaining unexpired patents are usable on a royalty-free basis.<br>
> https://en.wikipedia.org/wiki/G.729.1
https://git.jami.net/savoirfairelinux/ring-daemon/issues/200Backloghttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1054Android SIP client not working after update to Taranis2022-11-11T18:35:52ZFietzeAndroid SIP client not working after update to TaranisBug report form
---------------
## Environment
Version: Taranis 20211210-01
Device model: Nexus 5X
Network connection: WiFi FritzBox 7530
Android version: LineageOS 17.1 (Android 10)
Build: F-droid
## Steps to reproduce
1...Bug report form
---------------
## Environment
Version: Taranis 20211210-01
Device model: Nexus 5X
Network connection: WiFi FritzBox 7530
Android version: LineageOS 17.1 (Android 10)
Build: F-droid
## Steps to reproduce
1. tap on app logo to open Jami application
2. choose SIP account
3. tap on any of the existing conversations
4. conversations opens, showing the call history
5. tap on receiver icon
- Actual result: Screen turns dark grey. After a few second it falls back to the call history, showing "Missed outgoing call"
Another few seconds later, screen switches back to the conversations list.
Sometimes an error message appears "Jami keeps stopping"
Quite often, after some seconds Jami just closes down.
- Expected result: Jami should call the requested contact; I should hear a dailing tone.
## Additional information
SIP provider: sip.diamondcard.us
On 14th of November 2021 I had made a successful call of 10 min, 35 secs
I have not deliberately changed any of the settings - not the router nor the phone.
To verify the devices function, I have installed and set up the SIPdroid app. It works like a charm.https://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/40ffmpeg: add G711, G729 codecs (SIP)2022-02-03T18:28:56Zovariffmpeg: add G711, G729 codecs (SIP)Please add the following codecs to Jami:
* [G.711a](https://en.wikipedia.org/wiki/G.711)
* [G.711u](https://en.wikipedia.org/wiki/G.711)
* [G.729a](https://en.wikipedia.org/wiki/G.729)
What do you think?
Thank you
> MyNetFone uses th...Please add the following codecs to Jami:
* [G.711a](https://en.wikipedia.org/wiki/G.711)
* [G.711u](https://en.wikipedia.org/wiki/G.711)
* [G.729a](https://en.wikipedia.org/wiki/G.729)
What do you think?
Thank you
> MyNetFone uses the **G.711a** codec, it is the standard codec used in Australia and Europe. The **G.711u** codec is used within the US, and may be present in PBXs or UC platforms from US vendors. The **G.729a** codec is often used for conference calling services.<br>
> https://www.mynetfone.com.au/support/faq/question/477
> The patents for G.711, released in 1972, have expired, so it may be used without the need for a licence
> https://en.wikipedia.org/wiki/G.711#Licensing
> As of January 1, 2017, the patent terms of most licensed patents under the G.729 Consortium have expired, the remaining unexpired patents are usable on a royalty-free basis.<br>
> https://en.wikipedia.org/wiki/G.729.1
https://git.jami.net/savoirfairelinux/ring-daemon/issues/200Backloghttps://git.jami.net/savoirfairelinux/jami-product-backlog/-/issues/12Support DTMF2022-02-16T21:37:18ZSébastien BlinSupport DTMFTo be definedTo be definedBackloghttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/662Support DTMF2021-12-29T17:07:16ZSébastien BlinSupport DTMFTo be definedTo be definedBackloghttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1021SIP messaging failure causes authetication failure2022-11-11T16:34:31Zarkanoid87SIP messaging failure causes authetication failureI'm aware that this could be a ticket that could really be multiple ones, but I'm not sure what's causing this and it really feels like an uroboro.
I've an asterisk server running pjsip module to provide SIP calls and messages to vpn co...I'm aware that this could be a ticket that could really be multiple ones, but I'm not sure what's causing this and it really feels like an uroboro.
I've an asterisk server running pjsip module to provide SIP calls and messages to vpn connected clients (no direct connections). I'm already successfully using it with other Android clients like linphone and zoiper but I'm finding issues with Jami.
- Jami version: Maloya - 20211004-01 (Play Store)
- Device model: Samsung Note 10+
- Android version: 11
How to reproduce:
1) I setup my sip account in Jami -> authentication successful
2) I call myself or others -> call works
3) I try to send a message to myself or others -> no message is delivered, no sip packages are received on the server, app goes in "connection error" state.
4) I try to call call me again -> 401 not authorized
5) I try to enable and disable sip account -> 401 not authorized
6) Open menu "Account > advanced > Local Interface" and re-select already selected option (tun0 in my case) -> back to step 2
There are alternative steps "6" to rollback to step 2, like enabling and disabling "Published same as local"https://git.jami.net/savoirfairelinux/jami-project/-/issues/1329Settings: move use STUN/Stun address into SIP Account2022-11-03T02:34:32ZSébastien BlinSettings: move use STUN/Stun address into SIP Accountas it's unnecessary for Jami accounts (we have the DHT)as it's unnecessary for Jami accounts (we have the DHT)https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1020Allow adding a SIP account without creating a Jami user first2022-10-21T11:49:18Zarkanoid87Allow adding a SIP account without creating a Jami user firstI've found Jami inserted in https://en.wikipedia.org/wiki/List_of_SIP_software#Mobile_clients but on first download I've been quite surprised to have to create jami user, add sip account, delete jami user, to make it workI've found Jami inserted in https://en.wikipedia.org/wiki/List_of_SIP_software#Mobile_clients but on first download I've been quite surprised to have to create jami user, add sip account, delete jami user, to make it workhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/654"SEGV on unknown address" For SIP2021-10-27T21:56:02Zvindicatorr"SEGV on unknown address" For SIPI built and installed master of everything a moment ago.
I ran jami-qt and enabled my SIP account I had previously set up (not sure I ever actually connected before from previous tests).
This caused the crash.
After everything sett...I built and installed master of everything a moment ago.
I ran jami-qt and enabled my SIP account I had previously set up (not sure I ever actually connected before from previous tests).
This caused the crash.
After everything settled, I manually ran:
```
$ /usr/local/libexec/jamid -cd
Jami Daemon 10.1.0-68f1732948, by Savoir-faire Linux 2004-2019
https://jami.net/
[Video support enabled]
[Plugins support enabled]
[1634724962.374|16769|ringbuffer.cpp :55 ] Create new RingBuffer audiolayer_id
[1634724962.375|16769|manager.cpp :727 ] Not initialized
[1634724962.375|16769|manager.cpp :727 ] Not initialized
05:16:02.392 os_core_unix.c !pjlib 2.11 for POSIX initialized
[1634724962.392|16769|manager.cpp :798 ] Using PJSIP version 2.11 for x86_64-pc-linux-gnu
[1634724962.392|16769|manager.cpp :799 ] Using GnuTLS version 3.7.2
[1634724962.392|16769|manager.cpp :800 ] Using OpenDHT version 2.3.0
[1634724962.392|16769|manager.cpp :801 ] Using FFmpeg version n4.4
[1634724962.392|16769|manager.cpp :804 ] Using Libgit2 version 1.2.0
[1634724962.393|16769|sipvoiplink.cpp :659 ] Using SIP nameserver: 1.1.1.1
[1634724962.393|16769|sipvoiplink.cpp :659 ] Using SIP nameserver: 9.9.9.9
[1634724962.393|16769|sipvoiplink.cpp :659 ] Using SIP nameserver: 8.8.8.8
[1634724962.394|16769|sipvoiplink.cpp :753 ] SIPVoIPLink@<sanitized>
[1634724962.394|16769|manager.cpp :820 ] Configuration file path: /home/username/.config/jami/dring.yml
[1634724962.397|16769|accel.cpp :206 ] -- Starting encoding init for cuda with default device.
[AVHWDeviceContext @ 0x609000000700] Cannot load libcuda.so.1
[AVHWDeviceContext @ 0x609000000700] Could not dynamically load CUDA
[1634724962.397|16769|accel.cpp :171 ] Failed to create cuda device: -1313558101.
[1634724962.397|16769|accel.cpp :221 ] -- Init failed for cuda with default device.
[1634724962.398|16769|accel.cpp :228 ] -- Init encoding for cuda with device 1.
[AVHWDeviceContext @ 0x609000000a40] Cannot load libcuda.so.1
[AVHWDeviceContext @ 0x609000000a40] Could not dynamically load CUDA
[1634724962.398|16769|accel.cpp :171 ] Failed to create cuda device: -1313558101.
[1634724962.398|16769|accel.cpp :240 ] -- Init failed for cuda with device 1.
[1634724962.398|16769|accel.cpp :228 ] -- Init encoding for cuda with device 2.
[AVHWDeviceContext @ 0x609000000dc0] Cannot load libcuda.so.1
[AVHWDeviceContext @ 0x609000000dc0] Could not dynamically load CUDA
[1634724962.398|16769|accel.cpp :171 ] Failed to create cuda device: -1313558101.
[1634724962.398|16769|accel.cpp :240 ] -- Init failed for cuda with device 2.
[1634724962.403|16769|accel.cpp :206 ] -- Starting encoding init for vaapi with default device.
[1634724962.406|16769|accel.cpp :182 ] Device type vaapi successfully created.
[1634724962.406|16769|accel.cpp :215 ] -- Init passed for vaapi with default device.
[hevc_vaapi @ 0x61900002cb80] No usable encoding profile found.
[1634724962.407|16769|media_encoder.cpp :1178 ] Fail to open hardware encoder H265 with vaapi
[1634724962.407|16769|system_codec_container.cpp:197 ] Can't find a usable accelerated H265/HEVC codec, disabling.
[1634724962.407|16769|system_codec_container.cpp:232 ] Encoders found: H264 VP8 MP4V-ES H263-1998 opus G722 speex speex speex PCMA PCMU
[1634724962.408|16769|system_codec_container.cpp:233 ] Decoders found: H264 VP8 MP4V-ES H263-1998 opus G722 speex speex speex PCMA PCMU
[1634724962.408|16769|sipaccount.cpp :2407 ] All audio codecs disabled, enabling all
[1634724962.408|16769|sipaccount.cpp :2411 ] All video codecs disabled, enabling all
[1634724962.408|16769|upnp_context.cpp :39 ] Creating UPnPContext instance [<sanitized>]
[1634724962.408|16769|upnp_control.cpp :39 ] Controller@<sanitized>: Created UPnP Controller session
[1634724962.408|16770|upnp_context.cpp :409 ] Successfully registered controller <sanitized>
[1634724962.408|16770|upnp_context.cpp :131 ] Starting UPNP context
[1634724962.408|16769|sipaccount.cpp :1818 ] Set SIP registration EXPIRE to 60 - current 3600
[1634724962.408|16769|sipaccount.cpp :1870 ] Presence enabled for <sanitized> : false.
[1634724962.410|16774|jamiaccount.cpp :327 ] [Account <sanitized>] Can't load proxy URL from cache: Can't check write time for: /home/username/.cache/jami/<sanitized>/dhtproxy
[1634724962.414|16774|jamiaccount.cpp :1101 ] [Account <sanitized>] loading account
[1634724962.414|16774|namedirectory.cpp :475 ] Could not load /home/username/.cache/jami/namecache/ns.jami.net
[1634724962.414|16774|account_manager.cpp:57 ] Loading certificate from 'ring_device.crt' and key from 'ring_device.key' at /home/username/.local/share/jami/<sanitized>
[1634724962.434|16774|certstore.cpp :89 ] CertificateStore: loaded 36 local certificates.
[1634724962.436|16774|account_manager.cpp:143 ] [Auth] checking device receipt for <sanitized>
[1634724962.440|16774|contact_list.cpp :503 ] [Contacts] Found account device: computername <sanitized>
[1634724962.442|16774|contact_list.cpp :503 ] [Contacts] Found account device: <sanitized>
[1634724962.443|16774|account_manager.cpp:193 ] [Auth] Device <sanitized> receipt checked successfully for account <sanitized>
[1634724962.443|16774|jamiaccount.cpp :1207 ] [Account <sanitized>] loaded account identity
[1634724962.443|16774|conversation_module.cpp:678 ] [Account <sanitized>] Start loading conversations…
[1634724962.443|16774|conversation_module.cpp:1579 ] [convInfo] error loading convInfo: Can't read file: convInfo
[1634724962.443|16774|conversation_module.cpp:715 ] [Account <sanitized>] Conversations loaded!
[1634724962.443|16774|conversation_module.cpp:678 ] [Account <sanitized>] Start loading conversations…
[1634724962.443|16774|conversation_module.cpp:1579 ] [convInfo] error loading convInfo: Can't read file: convInfo
[1634724962.443|16774|conversation_module.cpp:715 ] [Account <sanitized>] Conversations loaded!
[1634724962.444|16769|ringbuffer.cpp :55 ] Create new RingBuffer urgentRingBuffer_id
[1634724962.447|16769|pulselayer.cpp :141 ] Waiting....
[1634724962.447|16779|pulselayer.cpp :141 ] Waiting....
[1634724962.447|16779|pulselayer.cpp :141 ] Waiting....
[1634724962.449|16779|pulselayer.cpp :145 ] Connection to PulseAudio server established
[1634724962.449|16779|pulselayer.cpp :175 ] Updating PulseAudio sink list
[1634724962.449|16779|pulselayer.cpp :192 ] Updating PulseAudio source list
[1634724962.449|16779|pulselayer.cpp :209 ] Updating PulseAudio server infos
[1634724962.451|16779|pulselayer.cpp :664 ] PulseAudio server info:
Server name: PulseAudio (on PipeWire 0.3.38)
Server version: 15.0.0
Default Sink bluez_output.00_02_76_64_A2_CF.a2dp-sink
Default Source alsa_input.pci-0000_00_1b.0.analog-stereo
Default Sample Specification: float32le 2ch 48000Hz
Default Channel Map: front-left,front-right
[1634724962.451|16779|audiolayer.cpp :64 ] Hardware audio format available : {s16, 2 channels, 48000Hz} 0
[1634724962.451|16779|manager.cpp :2870 ] Audio format changed: {s16, 1 channels, 16000Hz} -> {s16, 2 channels, 48000Hz}
[1634724962.496|16769|sipaccount.cpp :903 ] doRegister <sipserver>
[1634724962.496|16769|sipaccount.cpp :907 ] UPnP: waiting for IGD to register SIP account
[1634724962.496|16769|upnp_context.cpp :309 ] Try to find mapping for port 5060 [UDP]
[1634724962.496|16769|upnp_context.cpp :341 ] Did not find any available mapping. Will request one now
[1634724962.496|16769|upnp_context.cpp :1137 ] No IGD available. Mapping will be requested when an IGD becomes available
[1634724962.496|16769|mapping.cpp :86 ] Changing mapping JAMI-UDP:5060 state from AVAILABLE to UNAVAILABLE
[1634724962.496|16769|sipaccount.cpp :835 ] [Account <sanitized>] Failed to open port 5060: registering SIP account anyway
[1634724962.496|16769|sipvoiplink.cpp :1491 ] try to resolve '<sipserver>' (port: 0)
[1634724962.496|16769|sipaccount.cpp :910 ] UPnP: UPNP request failed, try to register SIP account anyway
[1634724962.496|16769|sipvoiplink.cpp :1491 ] try to resolve '<sipserver>' (port: 0)
[1634724962.497|16769|jamiaccount.cpp :1763 ] [Account <sanitized>] Starting account..
[1634724962.497|16769|jamiaccount.cpp :2599 ] [Account <sanitized>] connecting…
[1634724962.497|16770|upnp_context.cpp :622 ] UPNP/NAT-PMP enabled, but no valid IGDs available
[1634724962.497|16775|jamiaccount.cpp :2779 ] Loading DhParams from file '/home/username/.cache/jami/<sanitized>/dhParams'
[1634724962.497|16770|jamiaccount.cpp :1953 ] [Account <sanitized>] Starting account...
[1634724962.500|16770|jamiaccount.cpp :1794 ] [Account <sanitized>] Bootstrap node: bootstrap.jami.net
[1634724962.502|16782|jamiaccount.cpp :2075 ] [Account <sanitized>] Dht status: IPv4 connecting; IPv6 disconnected
[1634724962.616|16782|jamiaccount.cpp :2075 ] [Account <sanitized>] Dht status: IPv4 connected; IPv6 disconnected
[1634724962.616|16782|jamiaccount.cpp :2595 ] [Account <sanitized>] connected
[1634724962.617|16782|jamiaccount.cpp :3554 ] [Account <sanitized>] Store DHT public IPv4 address : <sanitized>
[1634724962.669|16773|sipaccount.cpp :1005 ] Creating transport
[1634724962.670|16773|siptransport.cpp :329 ] Created UDP transport on address 0.0.0.0:5060
[1634724962.670|16773|siptransport.cpp :80 ] SipTransport@<sanitized> {tr=<sanitized> {rc=2}}
[1634724962.670|16773|sipaccount.cpp :1099 ] Using contact header "SIP Provider" <sip:<sanitized>@<deviceip>:5060> in registration
[1634724962.697|16782|jamiaccount.cpp :2075 ] [Account <sanitized>] Dht status: IPv4 connected; IPv6 connecting
[1634724962.733|16773|sipaccount.cpp :1599 ] Using published address <sanitized> and port 5060
AddressSanitizer:DEADLYSIGNAL
=================================================================
==278913==ERROR: AddressSanitizer: SEGV on unknown address 0x000000000048 (pc 0x7f7fb357255b bp 0x7f7f9a9f7ae0 sp 0x7f7f9a9f7590 T4)
==278913==The signal is caused by a READ memory access.
==278913==Hint: address points to the zero page.
#0 0x7f7fb357255b in jami::SIPAccount::checkNATAddress(pjsip_regc_cbparam*, pj_pool_t*) (/usr/local/lib/libring.so.0+0x9f355b)
#1 0x7f7fb357b670 in jami::SIPAccount::onRegister(pjsip_regc_cbparam*) (/usr/local/lib/libring.so.0+0x9fc670)
#2 0x7f7fb3d6bd38 in regc_tsx_callback (/usr/local/lib/libring.so.0+0x11ecd38)
#3 0x7f7fb3d97256 in tsx_set_state (/usr/local/lib/libring.so.0+0x1218256)
#4 0x7f7fb3d985f7 in tsx_on_state_proceeding_uac (/usr/local/lib/libring.so.0+0x12195f7)
#5 0x7f7fb3d98825 in tsx_on_state_calling (/usr/local/lib/libring.so.0+0x1219825)
#6 0x7f7fb3d9ac9d in pjsip_tsx_recv_msg (/usr/local/lib/libring.so.0+0x121bc9d)
#7 0x7f7fb3d9ada4 in mod_tsx_layer_on_rx_response (/usr/local/lib/libring.so.0+0x121bda4)
#8 0x7f7fb3d845c6 in pjsip_endpt_process_rx_data (/usr/local/lib/libring.so.0+0x12055c6)
#9 0x7f7fb3d84785 in endpt_on_rx_msg (/usr/local/lib/libring.so.0+0x1205785)
#10 0x7f7fb3d8b7e8 in pjsip_tpmgr_receive_packet (/usr/local/lib/libring.so.0+0x120c7e8)
#11 0x7f7fb3d8e1ed in udp_on_read_complete (/usr/local/lib/libring.so.0+0x120f1ed)
#12 0x7f7fb3dd19d6 in ioqueue_dispatch_read_event (/usr/local/lib/libring.so.0+0x12529d6)
#13 0x7f7fb3dd343a in pj_ioqueue_poll (/usr/local/lib/libring.so.0+0x125443a)
#14 0x7f7fb3d842af in pjsip_endpt_handle_events2 (/usr/local/lib/libring.so.0+0x12052af)
#15 0x7f7fb35f5cf7 in std::thread::_State_impl<std::thread::_Invoker<std::tuple<jami::SIPVoIPLink::SIPVoIPLink()::{lambda()#1}> > >::_M_run() (/usr/local/lib/libring.so.0+0xa76cf7)
#16 0x7f7faf5f13c3 in execute_native_thread_routine /build/gcc/src/gcc/libstdc++-v3/src/c++11/thread.cc:82
#17 0x7f7fb260e258 in start_thread (/usr/lib/libpthread.so.0+0x9258)
#18 0x7f7faf2f15e2 in __GI___clone (/usr/lib/libc.so.6+0xfe5e2)
AddressSanitizer can not provide additional info.
SUMMARY: AddressSanitizer: SEGV (/usr/local/lib/libring.so.0+0x9f355b) in jami::SIPAccount::checkNATAddress(pjsip_regc_cbparam*, pj_pool_t*)
Thread T4 created by T0 here:
#0 0x7f7fb4218fa7 in __interceptor_pthread_create /build/gcc/src/gcc/libsanitizer/asan/asan_interceptors.cpp:216
#1 0x7f7faf5f16aa in std::thread::_M_start_thread(std::unique_ptr<std::thread::_State, std::default_delete<std::thread::_State> >, void (*)()) /build/gcc/src/gcc-build/x86_64-pc-linux-gnu/libstdc++-v3/include/x86_64-pc-linux-gnu/bits/gthr-default.h:663
#2 0x7f7fb2f244a2 in jami::Manager::init(std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char> > const&) (/usr/local/lib/libring.so.0+0x3a54a2)
==278913==ABORTING
```Mohamed ChibaniMohamed Chibanihttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1289Bundle same contacts with different spelling2021-08-19T19:54:53ZmokkinBundle same contacts with different spellingThe following examples shows three different spellings of the same contact:
![image](/uploads/564ff72a0adad7f06a3055ba20dd53f6/image.png)
1. International format with country code and city code
2. National format with city code
3. Local...The following examples shows three different spellings of the same contact:
![image](/uploads/564ff72a0adad7f06a3055ba20dd53f6/image.png)
1. International format with country code and city code
2. National format with city code
3. Local format without any prefix
All of them are successful for calling, because the pbx/sip knows its region. For a better overview these contacts should be recognized as the same and bundled.https://git.jami.net/savoirfairelinux/jami-daemon/-/issues/572Outgoing SIP calls drop after 32s2021-11-26T15:15:26ZDavid LecompteOutgoing SIP calls drop after 32sI am using an OVH SIP account. Outgoing calls to a landline or to another OVH SIP account drop after 32s, this is 100% reproducible.
I am using Trisquel 9 on a desktop PC and added Jami repository, the version of jami is 2021-03-08 21:4...I am using an OVH SIP account. Outgoing calls to a landline or to another OVH SIP account drop after 32s, this is 100% reproducible.
I am using Trisquel 9 on a desktop PC and added Jami repository, the version of jami is 2021-03-08 21:43:25 UTC . The network connection is via ethernet, my ISP is SFR in Paris France, fiber connection.
The attached log was obtained with
/usr/lib/ring/dring -d -c 2>&1 | tee dring5.log
The image is a screen capture taken immediately after the call drop (less than 1s after) in order to show what was visible in the log. The big seqence of consecutive "underrun occured" was during the call but it paused more than 10s before the call drop.
[dring5.log](/uploads/648d1b648fb16fc641573235420d2b37/dring5.log)
![right_after_drop](/uploads/6f53ee76e0969c94a429aac2589da7e8/right_after_drop.png)Mohamed ChibaniMohamed Chibanihttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/966Sip no audio for the called party2022-10-23T21:34:54ZkrishnakSip no audio for the called partySIP server has been tested well with other SIP clients - the new entry is only Jami
Jami registers on to SIP - call gets established (between JAMI and a PSTN), Jami user can hear the person on the other end. Called party (PSTN) doesn't ...SIP server has been tested well with other SIP clients - the new entry is only Jami
Jami registers on to SIP - call gets established (between JAMI and a PSTN), Jami user can hear the person on the other end. Called party (PSTN) doesn't hear Jami user.
Please note Jami to Jami audio and video calls work using the same Android devices.Ming Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/377Bugfix - SIP authentication username option missing in desktop clients2022-07-04T18:39:37ZRobinBugfix - SIP authentication username option missing in desktop clientsI'd like to refer to my request on [Twitter](https://twitter.com/jami_social/status/1217490812847763456)... BTW thanks for the quick reply :thumbsup:
---
I'm unable to set up my SIP account since there is no option for `authentication...I'd like to refer to my request on [Twitter](https://twitter.com/jami_social/status/1217490812847763456)... BTW thanks for the quick reply :thumbsup:
---
I'm unable to set up my SIP account since there is no option for `authentication username`. I have something like this:
```config
username: 1234
registrar: example.net
authentication username: johndoe
password: ***
```
Could you please fix this?Ming Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1224re-register to late2023-03-13T13:31:51ZLukas Wallischre-register to lateAs the logs suggests the re-register timer is to long... as far as i know the reregister should come before the old register expires, not 10s after...
```
[1618330072.037|43625|sipaccount.cpp :958 ] Start keep alive timer for accoun...As the logs suggests the re-register timer is to long... as far as i know the reregister should come before the old register expires, not 10s after...
```
[1618330072.037|43625|sipaccount.cpp :958 ] Start keep alive timer for account 7e9411aaf521bc4b
[1618330072.037|43625|sipaccount.cpp :977 ] Registration Expire: 119
[1618330072.037|43625|sipvoiplink.cpp :776 ] Register new keep alive timer 2099486383 with delay 129
```
OS : Ubuntu 20.04
Version: snap ( downloaded today)
`jami -v` gives:
```
Testing for explicit PulseAudio choice...
Testing for ALSA permissions...
...and using ALSA.
Jami Daemon 9.8.0-a6d5ad32d7-dirty, by Savoir-faire Linux 2004-2019
https://jami.net/
[Video support enabled]
[Plugins support enabled]
```
i can only find the settings for the expire-timer, but the not the reregister-timer. am I missing something?
I'm using this client behind a Hardware-PBX, because of the late reregister my calls keep getting abortedSébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1215Please detect status 488 from SIP server and offer suggestions to change codec2023-05-29T13:37:07ZreubenfirminPlease detect status 488 from SIP server and offer suggestions to change codecWhen a SIP server returns 488, it can mean a codec incompatibility. Your UI buries that 488 is returned. Please detect the status and pop up the settings asking the user to check the codecs vs what the voip provider allows.When a SIP server returns 488, it can mean a codec incompatibility. Your UI buries that 488 is returned. Please detect the status and pop up the settings asking the user to check the codecs vs what the voip provider allows.https://git.jami.net/savoirfairelinux/jami-client-gnome/-/issues/1257Outgoing SIP calls drop after 32s2021-06-25T19:17:17ZDavid LecompteOutgoing SIP calls drop after 32sI am using an OVH SIP account. Outgoing calls to a landline or to another OVH SIP account drop after 32s, this is 100% reproducible.
I am using Trisquel 9 on a desktop PC and added Jami repository, the version of jami is 2021-03-08 21:4...I am using an OVH SIP account. Outgoing calls to a landline or to another OVH SIP account drop after 32s, this is 100% reproducible.
I am using Trisquel 9 on a desktop PC and added Jami repository, the version of jami is 2021-03-08 21:43:25 UTC . The network connection is via ethernet, my ISP is SFR in Paris France, fiber connection.
The attached log was obtained with
/usr/lib/ring/dring -d -c 2>&1 | tee dring5.log
The image is a screen capture taken immediately after the call drop (less than 1s after) in order to show what was visible in the log. The big seqence of consecutive "underrun occured" was during the call but it paused more than 10s before the call drop.
[dring5.log](/uploads/13531e3ba91c41ce9edd1c87531a616b/dring5.log)
![right_after_drop](/uploads/32eccf95fdc4c7191b6c18ae7bab4d45/right_after_drop.png)Ming Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-client-gnome/-/issues/1255[GNOME] Dialpad2023-05-26T13:59:32ZJami Bot[GNOME] DialpadIssue generated from Tuleap's migration script.
**Originally submitted by: Óvári (ovari)**
Please add a Dialpad which has numbers (0,1,2,3,4,5,6,7,8,9) and star/asterisk (\*) and hash (\#).
https://support.skype.com/en/faq/FA22/how...Issue generated from Tuleap's migration script.
**Originally submitted by: Óvári (ovari)**
Please add a Dialpad which has numbers (0,1,2,3,4,5,6,7,8,9) and star/asterisk (\*) and hash (\#).
https://support.skype.com/en/faq/FA22/how-do-i-dial-an-extension-during-a-call
Thank you
ring-gnome 2018-03-23 23:25:11 UTC
Linux Mint 18.3 Cinnamon 64-bitMing Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/300Swarm: Remove SIP contact => crash2021-03-26T18:57:25ZSébastien BlinSwarm: Remove SIP contact => crash```
#0 0x00007fffee236d74 in QVariant::QVariant(QString const&) () at /lib64/libQt5Core.so.5
#1 0x000000000044aee2 in SmartListModel::getConversationItemData(lrc::api::conversation::Info const&, lrc::api::account::Info const&, int) con...```
#0 0x00007fffee236d74 in QVariant::QVariant(QString const&) () at /lib64/libQt5Core.so.5
#1 0x000000000044aee2 in SmartListModel::getConversationItemData(lrc::api::conversation::Info const&, lrc::api::account::Info const&, int) const ()
#2 0x0000000000449db0 in SmartListModel::data(QModelIndex const&, int) const ()
#3 0x00007fffef187acd in QQmlDMAbstractItemModelData::value(int) const () at /lib64/libQt5QmlModels.so.5
#4 0x00007fffef181f54 in QQmlDMCachedModelData::metaCall(QMetaObject::Call, int, void**) () at /lib64/libQt5QmlModels.so.5
#5 0x00007fffee8789a6 in loadProperty(QV4::ExecutionEngine*, QObject*, QQmlPropertyData const&) () at /lib64/libQt5Qml.so.5
#6 0x00007fffee879b17 in QV4::QObjectWrapper::getQmlProperty(QV4::ExecutionEngine*, QQmlContextData*, QObject*, QV4::String*, QV4::QObjectWrapper::RevisionMode, bool*, QQmlPropertyData**) () at /lib64/libQt5Qml.so.5
#7 0x00007fffee85f664 in QV4::QQmlContextWrapper::lookupInParentContextHierarchy(QV4::Lookup*, QV4::ExecutionEngine*, QV4::Value*) () at /lib64/libQt5Qml.so.5
```
# Scenario
+ Sip account add a contact
+ Right click
+ remove contactSwarm-chatAlbert Babí OllerAlbert Babí Oller