Hello all, My apologies if this has been asked other times but I have not found a working solution in any of the replies.
My Linphone setup consists of connecting to 1 PBX (asterisk) and registering 1 SIP extension on it. The registration is successful, placing and receiving calls works great and there are no other issues I can find with the app. The issue is that every time the app starts and then again every so often while it is running, a window titled "Unable to authenticate" pops up. Some posts suggest that some account is misconfigured but the only SIP account I have setup is working fine. Disabling the listening ports doesn't seem to have any effect. There is some aspect of the config that I believe I am missing. Version is 4.2.5 and I see the issue on Linux, Mac and Windows. Attached a linphonerc. Any suggestions are greatly appreciated. Thanks. Joao
[misc] version_check_url_root=https://linphone.org/releases log_collection_upload_server_url=https://www.linphone.org:444/lft.php uuid=3c2af981-1719-129f-ce6d-a63e56f239df transient_provisioning=1 [sound] ec_filter=MSWebRTCAEC remote_ring=/tmp/.mount_LinphopYnE1b/usr/share/sounds/linphone/ringback.wav playback_gain_db=0.000000 mic_gain_db=0.000000 ringer_dev_id=PulseAudio: Built-in Audio Analog Stereo playback_dev_id=PulseAudio: Plantronics .Audio 478 USB Digital Stereo (IEC958) capture_dev_id=PulseAudio: Plantronics .Audio 478 USB Analog Stereo [net] nat_policy_ref=L7To6GwLrMz0vGA mtu=1300 [nat_policy_0] ref=L7To6GwLrMz0vGA [sip] root_ca=/tmp/.mount_LinphopYnE1b/usr/share/linphone/rootca.pem verify_server_certs=1 verify_server_cn=1 contact=sip:user@unknown-host media_encryption=srtp guess_hostname=1 inc_timeout=30 in_call_timeout=0 delayed_timeout=4 register_only_when_network_is_up=1 register_only_when_upnp_is_ok=1 default_proxy=0 media_encryption_mandatory=1 sip_port=0 sip_tcp_port=0 sip_tls_port=-1 ipv6_migration_done=1 [app] auto_download_incoming_files_max_size=-1 sender_name_hidden_in_forward_message=0 [video] displaytype=MSOGL capture=1 display=1 size=vga preset=custom framerate=10.00000 device=V4L2: /dev/video0 [ui] rc_version=1 logs_enabled=1 [audio_codec_0] mime=opus rate=48000 channels=2 enabled=16 [audio_codec_1] mime=speex rate=16000 channels=1 enabled=0 [audio_codec_2] mime=speex rate=8000 channels=1 enabled=0 [audio_codec_3] mime=PCMU rate=8000 channels=1 enabled=16 [audio_codec_4] mime=PCMA rate=8000 channels=1 enabled=16 [audio_codec_5] mime=GSM rate=8000 channels=1 enabled=16 [audio_codec_6] mime=G722 rate=8000 channels=1 enabled=16 [audio_codec_7] mime=iLBC rate=8000 channels=1 enabled=0 [audio_codec_8] mime=G729 rate=8000 channels=1 enabled=16 [audio_codec_9] mime=speex rate=32000 channels=1 enabled=0 [audio_codec_10] mime=BV16 rate=8000 channels=1 enabled=0 [audio_codec_11] mime=L16 rate=44100 channels=2 enabled=0 [video_codec_0] mime=VP8 rate=90000 enabled=16 [rtp] audio_rtp_port=7078 video_rtp_port=9078 text_rtp_port=11078 audio_jitt_comp=60 video_jitt_comp=60 nortp_timeout=30 audio_adaptive_jitt_comp_enabled=1 video_adaptive_jitt_comp_enabled=1 [video_codec_2] mime=H265 rate=90000 enabled=0 [proxy_default_values] avpf=0 dial_escape_plus=0 publish=0 quality_reporting_enabled=0 quality_reporting_interval=0 reg_expires=3600 reg_sendregister=1 [assistant] algorithm=MD5 password_max_length=-1 password_min_length=0 username_length=-1 username_max_length=128 username_min_length=1 username_regex=^[a-zA-Z0-9+_.\-]*$ [nat_policy_1] ref=Karlm2m44-KebmQ stun_server=stun.mydomain.com:19302 protocols=stun,ice [auth_info_0] username=999 passwd=somepassword realm=asterisk domain=pbx.domain.com algorithm=md5 [proxy_0] reg_proxy=<sip:pbx.domain.com;transport=tls> reg_identity="Test account" <sip:[email protected]> quality_reporting_enabled=0 quality_reporting_interval=0 reg_expires=3600 reg_sendregister=1 publish=1 avpf=-1 avpf_rr_interval=1 dial_escape_plus=0 privacy=32768 push_notification_allowed=0 idkey=proxy_config_idkey publish_expires=3600 nat_policy_ref=Karlm2m44-KebmQ [audio_codec_12] mime=L16 rate=44100 channels=1 enabled=0
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