Здравствуйте.
Обновили Debian до:
No LSB modules are available.
Distributor ID: Debian
Description: Debian GNU/Linux 9.5 (stretch)
Release: 9.5
Codename: stretch
После этого раз в сутки отключается Flashphoner.
Логи отправил на почту.
Java Version:
openjdk version "1.8.0_181"
OpenJDK Runtime Environment (build 1.8.0_181-8u181-b13-1~deb9u1-b13)
OpenJDK 64-Bit Server VM (build 25.181-b13, mixed mode)
Конфиг:
	
	
	
		
								Обновили Debian до:
No LSB modules are available.
Distributor ID: Debian
Description: Debian GNU/Linux 9.5 (stretch)
Release: 9.5
Codename: stretch
После этого раз в сутки отключается Flashphoner.
Логи отправил на почту.
Java Version:
openjdk version "1.8.0_181"
OpenJDK Runtime Environment (build 1.8.0_181-8u181-b13-1~deb9u1-b13)
OpenJDK 64-Bit Server VM (build 25.181-b13, mixed mode)
Конфиг:
		Code:
	
	#Config
# ip                                - External IP-address of server where Flashphoner installed (xxx.xxx.xxx.xxx)
# ip_local                          - Local IP-address of server where Flashphoner installed (xxx.xxx.xxx.xxx)
# port_from                         - Begin of range of ports for SIP signaling (integer)
# port_to                           - End of range of ports for SIP signaling (integer)
# media_port_from                   - Begin of range of ports for media-traffic (integer)
# media_port_to                     - End of range of ports for media-traffic (integer)
# waiting_answer                    - How many time will Flashphoner client wait for answer from other side (seconds, integer)
# serial_number                     - Flashponer license. You can get it here - http://flashphoner.com/license
# user_agent                        - SIP User Agent header, example: Flashphoner/1.0 [Default: Flashphoner/1.0]
# balance_header                    - SIP header name for balance info [Default: balance]
# cost_header                       - SIP header name for cost info [Default: cost]
# video_enabled                     - Enabling of video support (true/false)
# domain                            - Domain address of voip server (xxx.xxx.xxx.xxx)
# outbound_proxy                    - Outbound proxy (xxx.xxx.xxx.xxx)
# outbound_port                     - Port for outbound proxy
# dtmf                              - Dual-Tone Multi-Frequency. Values: rfc2833, info
# log_level                         - Level of logging (1-10)
# enable_context_logs               - Context logs with login,port and sip call id
# rtp_activity_detecting            - Hangup, if incomming audio stream is empty few seconds. Format:(true/false),seconds. Example:true,5
# force_h264_to_sorenson            - Transcoding incoming stream from H264 to Sorenson Park
# force_h264_to_h264                - Transcoding incoming stream from H264 to H264 (frame without slices)
# h264_max_nalu_size                - Max NALU size for H264
# codecs                            - List of supported codecs, ordered by priority. Example: alaw, ulaw, g729, speex16, h263
# codecs_exclude_sip                - List of codecs that should be excluded in sip call
# codecs_exclude_streaming          - List of codecs that should be excluded in streaming
# codecs_exclude_sip_rtmp           - List of codecs that should be excluded in sip as rtmp
# sip_msg_listener                  - Class, defined by developer. The class must implement interface ISipMessageListener. Example: com.flashphoner.sdk.sip.ChangeCallIdListener
# priority_outside_codecs           - Priority of outside codecs
# remove_ssrc_attr                  - Remove ssrc attribute from local sdp
# max_callid_length                 - Max call id length (min 4; max 32)
# use_tcp_for_long_sip_messages     - Use TCP transport for messages with big size
# enable_candidate_harvester        - Enable stun candidate hasvester
# save_client_logs                  - Save client logs after disconnect
# enable_extended_logging           - Save user logs in individual folder, true is default value
# record                            - Folder for record audio from calls. If empty then recording will be disabled
# record_filename_template          - Template for file name of record. Default: {id}-{date}
# preserve_non_mixed_recorded_files - Preserve non mixed recorded files. Default: false
# recording_by_user                 - Recording media by user side. Default: false
ip                     =192.99.67.31
ip_local               =192.99.67.31
port_from              =30000
port_to                =31000
media_port_from        =31001
media_port_to          =32000
waiting_answer         =60
user_agent             =Flashphoner/1.0
balance_header         =balance
cost_header            =cost
video_enabled          =true
domain                 =
outbound_proxy         =
outbound_port          =
log_level              =5
suppress_audio         =true
enable_context_logs    =false
rtp_activity_detecting =true,60
sip_msg_listener       =com.flashphoner.sdk.sip.ChangeCallIdListener
call_record_listener   =com.flashphoner.server.client.DefaultCallRecordListener
dtmf                     =rfc2833
auto_login_url         =/usr/local/FlashphonerWebCallServer/conf/account.xml
get_callee_url         =/usr/local/FlashphonerWebCallServer/conf/callee.xml
codecs                   =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv
codecs_exclude_sip       =mpeg4-generic,flv,mpv
codecs_exclude_streaming =flv,telephone-event
codecs                   =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,vp8,h264,flv,mpv
on_record_hook_script  =on_record_hook.sh
rtmp_transponder_stream_name_prefix =rtmp_
webrtc_cc_min_bitrate = 500000
webrtc_cc_max_bitrate = 15000000
	
								
									Last edited: