CentOS release 6.7 (Final) Asterisk 11.17.0
На всякий приложу sip.conf и настройки оператора
[general]
allowguest=no
#include sip_trunk.conf         ; Там будут храниться настройки транков.
#include sip_sendcalls.conf     ; List accounts from sendcalls
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
allowtransfer=yes               ; Disable all transfers (unless enabled in peers or users)
udpbindaddr=0.0.0.0;             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
bindport=5060
transport=udp,tcp,tls
tcpenable=yes
tlsenable=yes
srvlookup=yes     ;              ; Enable DNS SRV lookups on outbound calls
toneduration=600
maxexpiry=7600                 ; Maximum allowed time of incoming registrations
minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
mwiexpiry=3600                 ; Expiry time for outgoing MWI subscriptions
disallow=all                   ; First disallow all codecs
allow=alaw                     ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g729
allow=h264
allow=h263                     ; see 
https://wiki.asterisk.org/wiki/display/ ... ketization
language=ru                    ; Default language setting for all users/peers;
dtmfmode = rfc2833             ; Set default dtmfmode for sending DTMF. Default: rfc2833
alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
dynamic_exclude_static = yes   ; Disallow all dynamic hosts from registering ; Запрещает регистрацию с динамичего Айпи.
rtptimeout=61                  ; Terminate call if 61 seconds of no RTP or RTCP activity
rtpholdtimeout=301             ; Terminate call if 301 seconds of no RTP or RTCP activity
allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
faxdetect=no
registertimeout=20              ; retry registration calls every 20 seconds (default)
registerattempts=0              ; Number of registration attempts before we give up
relaxdtmf=yes                   ; No sound from digit
dumphistory=yes;              ; вывод отчета по истории SIP в конце диалогового окна SIP. SIP-история записывается в канал протоколирования DEBUG
jbenable=yes
jbresyncthreshold=10000;         ; полезно для улучшения качества голоса, переданного со скачкообразными/прерывистыми временными метками, которые обычно пос$
jbimpl=fixed
jbmaxsize=20000
[link]
host=185.48.16.4
context=from-pstn
type=friend
insecure=port,invite
port=5060
transport=udp
qualify=yes
disallow=all
allow=alaw
;allow=g729
nat=no