Source: gnome-calls Version: 0.3.3-1 Severity: serious Tags: ftbfs https://buildd.debian.org/status/logs.php?pkg=gnome-calls&arch=amd64 https://buildd.debian.org/status/logs.php?pkg=gnome-calls&arch=all
... =================================== 14/16 ==================================== test: sip start time: 20:46:52 duration: 0.53s result: killed by signal 5 SIGTRAP command: MALLOC_CHECK_=2 G_TEST_BUILDDIR='/<<PKGBUILDDIR>>/_build/plugins/provider/tests' GSETTINGS_SCHEMA_DIR='/<<PKGBUILDDIR>>/_build/data' GSETTINGS_BACKEND=memory CALLS_AUDIOSRC=audiotestsrc G_TEST_SRCDIR='/<<PKGBUILDDIR>>/plugins/provider/tests' G_DEBUG=gc-friendly,fatal-warnings NO_AT_BRIDGE=1 CALLS_AUDIOSINK=fakesink MALLOC_PERTURB_=243 CALLS_SIP_TEST=1 PYTHONDONTWRITEBYTECODE=yes '/<<PKGBUILDDIR>>/_build/plugins/provider/tests/sip' ----------------------------------- stdout ----------------------------------- # random seed: R02Sad24a1244a453042551428c9a905d7e2 1..5 # Start of Calls tests # Start of SIP tests ok 1 /Calls/SIP/provider_object ok 2 /Calls/SIP/provider_origins # CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname # CallsSipOrigin-DEBUG: Account changed: # user: alice # host: x86-conova-01 # CallsSipMediaManager-DEBUG: Creating CallsSipMediaManager # GLib-GIO-DEBUG: _g_io_module_get_default: Found default implementation memory (GMemorySettingsBackend) for ?gsettings-backend? # CallsSettings-DEBUG: Setting country code to # CallsSettings-DEBUG: Enabling the use of default origins # CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available # CallsGstRfc3551-DEBUG: Adding PCMA to the codec candidates # CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMU is available # CallsGstRfc3551-DEBUG: Adding PCMU to the codec candidates # CallsGstRfc3551-DEBUG: Gstreamer plugin for GSM is available # CallsGstRfc3551-DEBUG: Adding GSM to the codec candidates # CallsGstRfc3551-DEBUG: Gstreamer plugin for G723 is not available # CallsSipMediaManager-DEBUG: Did not find audio codec G722 # CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available # CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMU is available # CallsGstRfc3551-DEBUG: Gstreamer plugin for GSM is available # CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname # CallsSipOrigin-DEBUG: Account changed: # user: bob # host: x86-conova-01 # CallsSipOrigin-DEBUG: Clearing any handles # CallsSipOrigin-DEBUG: Requesting nua_shutdown () # CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY # CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful # CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY # CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle # CallsSipOrigin-DEBUG: Clearing any handles # CallsSipOrigin-DEBUG: Requesting nua_shutdown () # CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful # CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle # CallsSipOrigin-DEBUG: Clearing any handles # CallsSipOrigin-DEBUG: Requesting nua_shutdown () # CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful # CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle ok 3 /Calls/SIP/origin_objects # CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname # CallsSipOrigin-DEBUG: Account changed: # user: alice # host: x86-conova-01 # CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname # CallsSipOrigin-DEBUG: Account changed: # user: bob # host: x86-conova-01 # CallsSipOrigin-DEBUG: Clearing any handles # CallsSipOrigin-DEBUG: Requesting nua_shutdown () # CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful # CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle # CallsSipOrigin-DEBUG: Clearing any handles # CallsSipOrigin-DEBUG: Requesting nua_shutdown () # CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful # CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle # CallsSipOrigin-DEBUG: Clearing any handles # CallsSipOrigin-DEBUG: Requesting nua_shutdown () # CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful # CallsSipOrigin-DEBUG: nua_shutdown() complete. Destroying nua handle ok 4 /Calls/SIP/origin_call_lists # CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname # CallsSipOrigin-DEBUG: Account changed: # user: alice # host: x86-conova-01 # CallsSipOrigin-DEBUG: Direct connection case. Using user and hostname # CallsSipOrigin-DEBUG: Account changed: # user: bob # host: x86-conova-01 # DEBUG: Call test: Stage 1 # CallsSipOrigin-DEBUG: Calling `sip:alice@127.0.0.1:5060' from origin 'bob' # CallsSipOrigin-DEBUG: Setting local SDP for outgoing call to sip:alice@127.0.0.1:5060: # v=0 # m=audio 53611 RTP/AVP 8 0 3 # a=rtpmap:8 PCMA/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:3 GSM/8000 # a=rtcp:52537 # # # CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY # CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY # CallsSipOrigin-DEBUG: The call state has changed: 000 INVITE sent # CallsSipOrigin-DEBUG: incoming call INVITE: 100 Trying from sip:bob@x86-conova-01 # CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY # CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY # DEBUG: Hanging up call # CallsSipCall-DEBUG: Hanging up incoming call # CallsSipOrigin-DEBUG: The call state has changed: 100 Trying # CallsSipOrigin-DEBUG: Found common codec: PCMA # CallsSipOrigin-DEBUG: Found common codec: PCMU # CallsSipOrigin-DEBUG: Found common codec: GSM # CallsSipOrigin-DEBUG: Remote SDP was set to: # v=0 # o=- 3562168267639768217 5101368152198370217 IN IP6 2a02:16a8:dc41:100::238 # s=- # c=IN IP6 2a02:16a8:dc41:100::238 # t=0 0 # m=audio 53611 RTP/AVP 8 0 3 # a=rtpmap:8 PCMA/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:3 GSM/8000 # a=rtcp:52537 # # CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 53611/52537 # CallsSipOrigin-DEBUG: Call incoming # CallsSipOrigin-DEBUG: The call state has changed: 480 Call state # CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline # CallsSipMediaPipeline-DEBUG: Stopping media pipeline # CallsSipMediaPipeline-DEBUG: Stopping media pipeline # DEBUG: Call test: Stage 2 # CallsSipOrigin-DEBUG: Calling `sip:bob@127.0.0.1:5061' from origin 'alice' # CallsSipOrigin-DEBUG: Setting local SDP for outgoing call to sip:bob@127.0.0.1:5061: # v=0 # m=audio 49100 RTP/AVP 8 0 3 # a=rtpmap:8 PCMA/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:3 GSM/8000 # a=rtcp:38085 # # # CallsSipOrigin-DEBUG: response to outgoing INVITE: 480 Temporarily Unavailable # CallsSipOrigin-DEBUG: The call state has changed: 480 Temporarily Unavailable # CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline # CallsSipMediaPipeline-DEBUG: Stopping media pipeline # CallsSipMediaPipeline-DEBUG: Stopping media pipeline # CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from READY to NULL # CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY # CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from READY to NULL # CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from READY to NULL # CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY # CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from READY to NULL # CallsSipOrigin-DEBUG: The call state has changed: 000 INVITE sent # CallsSipOrigin-DEBUG: incoming call INVITE: 100 Trying from sip:alice@x86-conova-01 # CallsSipOrigin-DEBUG: The call state has changed: 100 Trying # CallsSipOrigin-DEBUG: Found common codec: PCMA # CallsSipOrigin-DEBUG: Found common codec: PCMU # CallsSipOrigin-DEBUG: Found common codec: GSM # CallsSipOrigin-DEBUG: Remote SDP was set to: # v=0 # o=- 9211933157011057919 6832086157911654361 IN IP6 2a02:16a8:dc41:100::238 # s=- # c=IN IP6 2a02:16a8:dc41:100::238 # t=0 0 # m=audio 49100 RTP/AVP 8 0 3 # a=rtpmap:8 PCMA/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:3 GSM/8000 # a=rtcp:38085 # # CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 49100/38085 # CallsSipOrigin-DEBUG: Call incoming # CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from NULL to READY # CallsSipMediaPipeline-DEBUG: Element rtcp-udp-src has changed state from NULL to READY # DEBUG: Answering incoming call # CallsSipCall-DEBUG: Setting local SDP to string: # v=0 # m=audio 57548 RTP/AVP 8 0 3 # a=rtpmap:8 PCMA/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:3 GSM/8000 # a=rtcp:41973 # # # DEBUG: Hanging up call # CallsSipCall-DEBUG: Hanging up ongoing call # CallsSipOrigin-DEBUG: The call state has changed: 200 Call state # CallsSipOrigin-DEBUG: response to outgoing INVITE: 200 OK # CallsSipOrigin-DEBUG: The call state has changed: 200 OK # CallsSipOrigin-DEBUG: Found common codec: PCMA # CallsSipOrigin-DEBUG: Remote SDP was set to: # v=0 # o=- 1371930777682465956 1749628799533106578 IN IP6 2a02:16a8:dc41:100::238 # s=- # c=IN IP6 2a02:16a8:dc41:100::238 # t=0 0 # m=audio 57548 RTP/AVP 8 # a=rtpmap:8 PCMA/8000 # a=rtcp:41973 # # CallsSipCall-DEBUG: Setting remote ports: RTP/RTCP 57548/41973 # CallsSipOrigin-DEBUG: Call ready. Activating media pipeline # CallsSipCall-DEBUG: Setting codec 'PCMA' for pipeline # CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available # CallsSipMediaPipeline-DEBUG: Capabilities: # application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA,payload=(int)8 # CallsSipMediaPipeline-DEBUG: Starting media pipeline # CallsSipMediaPipeline-DEBUG: RTP/RTCP port before starting pipeline: 49100/38085 # CallsSipMediaPipeline-DEBUG: RTP/RTCP port after starting pipeline: 49100/38085 # CallsSipOrigin-DEBUG: incoming ACK: 200 OK # CallsSipOrigin-DEBUG: The call state has changed: 200 OK # CallsSipOrigin-DEBUG: Call ready. Activating media pipeline # CallsSipCall-DEBUG: Setting codec 'PCMA' for pipeline # CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available # CallsSipMediaPipeline-DEBUG: Capabilities: # application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA,payload=(int)8 # CallsSipMediaPipeline-DEBUG: Starting media pipeline # CallsSipMediaPipeline-DEBUG: RTP/RTCP port before starting pipeline: 57548/41973 # CallsSipMediaPipeline-DEBUG: RTP/RTCP port after starting pipeline: 57548/41973 # CallsSipOrigin-DEBUG: response to BYE: 200 OK # CallsSipOrigin-DEBUG: The call state has changed: 200 to BYE # CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline # CallsSipMediaPipeline-DEBUG: Stopping media pipeline # CallsSipMediaPipeline-DEBUG: Stopping media pipeline # CallsSipOrigin-DEBUG: incoming BYE: 200 Session Terminated # CallsSipOrigin-DEBUG: The call state has changed: 200 Session Terminated # CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline # CallsSipMediaPipeline-DEBUG: Stopping media pipeline # CallsSipMediaPipeline-DEBUG: Stopping media pipeline Bail out! CallsSipMediaPipeline-FATAL-WARNING: Error on the message bus: Could not get/set settings from/on resource. (../gst/udp/gstmultiudpsink.c(1228): gst_multiudpsink_configure_client (): /GstPipeline:media-pipeline/GstUDPSink:rtcp-udp-sink: Invalid address family (got 10)) ----------------------------------- stderr ----------------------------------- su_source_port_create() returns 0x558fc04f60c0 su_source_port_create() returns 0x558fc04eecc0 su_source_port_create() returns 0x558fc04f60c0 sres: /etc/resolv.conf: unknown option sres: /etc/resolv.conf: unknown option sres: /etc/resolv.conf: unknown option su_source_port_create() returns 0x558fc04f60c0 sres: /etc/resolv.conf: unknown option sres: /etc/resolv.conf: unknown option sres: /etc/resolv.conf: unknown option su_source_port_create() returns 0x558fc04f60c0 sres: /etc/resolv.conf: unknown option sres: /etc/resolv.conf: unknown option sres: /etc/resolv.conf: unknown option (/<<PKGBUILDDIR>>/_build/plugins/provider/tests/sip:3109934): CallsSipMediaPipeline-WARNING **: 20:46:52.860: Error on the message bus: Could not get/set settings from/on resource. (../gst/udp/gstmultiudpsink.c(1228): gst_multiudpsink_configure_client (): /GstPipeline:media-pipeline/GstUDPSink:rtcp-udp-sink: Invalid address family (got 10)) ============================================================================== ... Summary of Failures: 14/16 sip FAIL 0.53s killed by signal 5 SIGTRAP Ok: 15 Expected Fail: 0 Fail: 1 Unexpected Pass: 0 Skipped: 0 Timeout: 0 dh_auto_test: error: cd _build && LC_ALL=C.UTF-8 MESON_TESTTHREADS=4 meson test returned exit code 1 make[1]: *** [debian/rules:29: override_dh_auto_test] Error 25