Calls don't work with Asterisk

1

I see hint about incoming call or about outgoing call in window but real call doesn’t happens.

$ jami --version
Jami 202510031555

Kubuntu 24.04.3

And I don’t see any reason WHY audio call is stopped in log. WHYYYYYYYY?

But I see notification about iHD_drv_video while video calls switched off at all

$ jami
client  [1763017014565][INF][70b84ac66300]: Using Qt runtime version: 6.8.3
client  [1763017014570][INF][70b84ac66300]: "notify server name: Plasma, vendor: KDE, version: 5.27.12, spec: 1.2"
client  [1763017014575][INF][70b84ac66300]: "Using locale: en_US"
09:56:54.587         os_core_unix.c !pjlib 2.15.1 for POSIX initialized
Daemon is running
client  [1763017014781][INF][70b84ac66300]: Screen saver dbus interface:  "org.freedesktop.ScreenSaver"
qml     [1763017014791][INF][70b84ac66300]:[qrc:/MainApplicationWindow.qml:147] Initializing main view
default [1763017014791][WRN][70b84ac66300]: qrc:/MainApplicationWindow.qml:33:1: QML MainApplicationWindow: Conflicting properties 'visible' and 'visibility'
client  [1763017015014][INF][70b84ac66300]: Main window loaded using "OpenGLRhi"
default [1763017015014][WRN][70b84ac66300]: Loaded existing settings for account: "6ad27ee60d5647a8"
client  [1763017015187][INF][70b84ac66300]: NetworkManager client initialized, version: 1.46.0 , daemon running: yes , networking enabled: yes
client  [1763017015187][INF][70b84ac66300]: Primary network connection: bfeb0ca2-7c53-3596-90d5-56bd03fdd1a0 default: yes
[libopus @ 0x70b7c81d9780] Could not update timestamps for skipped samples.
[libopus @ 0x70b7c81d9780] Could not update timestamps for discarded samples.
libva info: VA-API version 1.20.0
libva info: Trying to open /usr/lib/x86_64-linux-gnu/dri/iHD_drv_video.so
libva info: Found init function __vaDriverInit_1_20
libva info: va_openDriver() returns 0
qml     [1763017019614][WRN][70b84ac66300]:[qrc:/mainview/components/CallsButton.qml:94] ACTIVE CALLS: false
2

Can you provide more information? I understand that your Jami is used as a SIP client regisitered on an Asterisk server. When you added a SIP account did you select RTP or TLS ? Also, check if you the SDES key exchange and TLS are enabled in your Jami SIP account’s advanced settings. If your asterisk server does not encrypt communications you must use RTP when you create the SIP account in Jami and ensure SDES key exchance and TLS are disabled in the advanced settings.

1 Like
3

I switched off SDES, TLS and changed preferred codecs, now calls from me works

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1 Like
4

Please advise any corrections or improvements to the https://review.jami.net/c/jami-docs/+/32395 patch.

Once merged, the information will be available at the following link.

Thank you

5

@QuAzI, can you please mark this topic as resolved with a solution? Thank you