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Why did the voice disappear during calls?
Hello. There is a main office where costs CUCM 8 + Asterisk + FreeBSD 10.3 + Cisco G7911. There are remote branches on them are Mikrotik RB951H-2nD + Cisco G7911.
The telephony internal between office and branches goes wonderfully. Ipsec has been raised from Mikrotika to FreeBSD. Nothing has changed and until recently there was a connection between the branches. Now there is signal traffic, but there is no voice (((
Already re-watched everything, but there is still no voice. Which way to dig?
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[general]
context=default
allowguest=no
;match_auth_username=yes
allowoverlap=no
externip=1.1.1.1
;allowoverlap=yes
;allowoverlap=dtmf
;allowtransfer=no
;realm=mydomain.tld
;domainsasrealm=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
;tlsenable=no
;tlsbindaddr=0.0.0.0
;tcpauthtimeout = 30
;tcpauthlimit = 100
transport=udp
srvlookup=no
;pedantic=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
tos_text=af41
;cos_sip=3
;cos_audio=5
;cos_video=4
;cos_text=3
;maxexpiry=3600
;minexpiry=60
;defaultexpiry=120
;mwiexpiry=3600
;maxforwards=70
;qualifyfreq=60
;qualifygap=100
;qualifypeers=1
;notifymimetype=text/plain
buggymwi=yes
;mwi_from=asterisk
vmexten=voicemail
;preferred_codec_only=yes
disallow=all
allow=ulaw
allow=alaw
;allow=ilbc
;mohinterpret=default
;mohsuggest=default
;parkinglot=plaza
language=ru
;relaxdtmf=yes
;trustrpid = no
;sendrpid = yes
;sendrpid = rpid
;sendrpid = pai
;rpid_update = no
;prematuremedia=no
;progressinband=never
;useragent=Asterisk PBX
;promiscredir = no
;usereqphone = no
;dtmfmode = rfc2833
;compactheaders = yes
videosupport=yes
;maxcallbitrate=384
;callevents=no
;authfailureevents=no
;alwaysauthreject = no
;auth_options_requests = yes
;g726nonstandard = yes
;outboundproxy=proxy.provider.domain
;outboundproxy=proxy.provider.domain:8080
;outboundproxy=proxy.provider.domain,force
;outboundproxy=tls://proxy.provider.domain
;outboundproxy=192.0.2.1
;outboundproxy=2001:db8::1
;outboundproxy=192.168.0.2.1:5062
;outboundproxy=[2001:db8::1]:5062
;matchexternaddrlocally = yes
;dynamic_exclude_static = yes
externaddr=1.1.1.1
contactdeny=0.0.0.0/0.0.0.0
contactpermit=192.168.11.0/255.255.255.0
contactpermit=192.168.12.0/255.255.255.0
contactpermit=192.168.13.0/255.255.255.0
contactpermit=192.168.14.0/255.255.255.0
contactpermit=192.168.16.0/255.255.255.0
contactpermit=192.168.19.0/255.255.255.0
contactpermit=192.168.240.0/255.255.255.0
;engine=asterisk
;regcontext=sipregistrations
;regextenonqualify=yes
;legacy_useroption_parsing=yes
;shrinkcallerid=yes
;use_q850_reason = no
;t1min=100
;timert1=500
;timerb=32000
rtptimeout=20
;rtpholdtimeout=300
;rtpkeepalive=<secs>
;session-timers=originate
;session-expires=600
;session-minse=90
;session-refresher=uas
;sipdebug = yes
;recordhistory=yes
;dumphistory=yes
;allowsubscribe=no
subscribecontext = default
notifyringing = no
notifyhold = yes
notifycid = yes
callcounter = yes
directmedia=no
;directmedia=nonat
;directmedia=update
;directrtpsetup=yes
directmediadeny=0.0.0.0/0
;directmediapermit=192.168.1.0/24
;ignoresdpversion=yes
;sdpsession=Asterisk PBX
;sdpowner=root
;encryption=no
;rtcachefriends=yes
;rtsavesysname=yes
;rtupdate=yes
;rtautoclear=yes
;ignoreregexpire=yes
counteronpeer=yes
localnet=192.168.12.0/255.255.255.
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