Your message dated Thu, 13 Oct 2022 01:49:09 +0000
with message-id <e1oinlr-006ovz...@fasolo.debian.org>
and subject line Bug#1019292: fixed in gnome-calls 43.0-2
has caused the Debian Bug report #1019292,
regarding gnome-calls FTBFS on IPV6-only buildds
to be marked as done.
This means that you claim that the problem has been dealt with.
If this is not the case it is now your responsibility to reopen the
Bug report if necessary, and/or fix the problem forthwith.
(NB: If you are a system administrator and have no idea what this
message is talking about, this may indicate a serious mail system
misconfiguration somewhere. Please contact ow...@bugs.debian.org
immediately.)
--
1019292: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=1019292
Debian Bug Tracking System
Contact ow...@bugs.debian.org with problems
--- Begin Message ---
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
--- End Message ---
--- Begin Message ---
Source: gnome-calls
Source-Version: 43.0-2
Done: Evangelos Ribeiro Tzaras <devrtz-deb...@fortysixandtwo.eu>
We believe that the bug you reported is fixed in the latest version of
gnome-calls, which is due to be installed in the Debian FTP archive.
A summary of the changes between this version and the previous one is
attached.
Thank you for reporting the bug, which will now be closed. If you
have further comments please address them to 1019...@bugs.debian.org,
and the maintainer will reopen the bug report if appropriate.
Debian distribution maintenance software
pp.
Evangelos Ribeiro Tzaras <devrtz-deb...@fortysixandtwo.eu> (supplier of updated
gnome-calls package)
(This message was generated automatically at their request; if you
believe that there is a problem with it please contact the archive
administrators by mailing ftpmas...@ftp-master.debian.org)
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA512
Format: 1.8
Date: Thu, 13 Oct 2022 00:36:08 +0200
Source: gnome-calls
Architecture: source
Version: 43.0-2
Distribution: unstable
Urgency: medium
Maintainer: DebianOnMobile Maintainers
<debian-on-mobile-maintain...@alioth-lists.debian.net>
Changed-By: Evangelos Ribeiro Tzaras <devrtz-deb...@fortysixandtwo.eu>
Closes: 1019292
Changes:
gnome-calls (43.0-2) unstable; urgency=medium
.
* d/patches: Fix build on IPv6 only buildd Closes: #1019292, thanks Adrian
Bunk
* d/patches: Reenable g722 codecs
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--- End Message ---
