История изменений [назад]

[general]  nat=yes  externip =
  xx.xx.xx.x =xx.xx.xx.x 
fromdomain =
  = xx.xxxxxxxxxxx.ru  localnet =
  192.168.X.0/255.255.255.0 qualify=yes
 
=192.168.X.0/255.255.255.0 
qualify=yes

context=default
allowoverlap=no               
bindaddr=192.168.0.100
udpbindaddr=192.168.0.100
tcpenable=yes
tcpbindaddr=192.168.0.100            
transport=udp
srvlookup=yes
tos_sip=cs3
defaultexpiry=360
disallow=all
allow=alaw
allow=g729
allow=723
allow=ulaw       
language=ru
dtmfmode = rfc2833
alwaysauthreject = yes
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=5
[1001] 
host=dynamic
context=default                 ;
  Default context for incoming calls.
  Defaults to 'default' ;allowguest=no 

  ; Allow or reject guest calls (default
  is yes) allowoverlap=no 

  ; Disable overlap dialing support.
  (Default is yes)
  bindaddr=192.168.0.100
  udpbindaddr=192.168.0.100            ;
  IP address to bind UDP listen socket
  to (0.0.0.0 binds to all)
  tcpenable=yes                   ;
  Enable server for incoming TCP
  connections (default is no)
  tcpbindaddr=192.168.0.100 

  ; IP address for TCP server to bind to
  (0.0.0.0 binds to all interfaces)
  transport=udp                   ; Set
  the default transports.  The order
  determines the primary default
  transport. srvlookup=yes 

  ; Enable DNS SRV lookups on outbound
  calls tossip=cs3                    ;
  Sets TOS for SIP packets. tosaudio=ef
  ; Sets TOS for RTP audio packets.
  defaultexpiry=360              ;
  Default length of incoming/outgoing
  registration
 
disallow=all                   ; First
  disallow all codecs allow=alaw
  allow=g729 allow=723 allow=ulaw 

  ; Allow codecs in order of preference
  language=ru                    ;
  Default language setting for all
  users/peers useragent=PBX-TEST 

  ; Allows you to change the user agent
  string dtmfmode = rfc2833 

  ; Set default dtmfmode for sending
  DTMF. Default: rfc2833 callevents=yes 
  ; generate manager events when sip ua
  alwaysauthreject = yes         ; When
  an incoming INVITE or REGISTER is to
  be rejected,
 
rtptimeout=60                  ;
  Terminate call if 60 seconds of no RTP
  or RTCP activity rtpholdtimeout=300 

  ; Terminate call if 300 seconds of no
  RTP or RTCP activity rtpkeepalive=5 

  ; Send keepalives in the RTP stream to
  keep NAT open [1001] host=dynamic
  context=default  type=friend
 username=1001  nat=yes
 secret=secret1001  callerid=phone1
  <1001> port=5060 insecure=invite
 

insecure=invite

[1002]  host=dynamic  context=default
 type=friend  username=1002  nat=yes
 secret=secret1002  callerid=phone2
  <1002> port=5060 insecure=invite
 

insecure=invite

extensions.conf

[general]  static=yes writeprotect=yes
 

writeprotect=yes

[default]  exten =>
  => 1001,1,Dial(SIP/1001)  exten =>
  1002,1,Dial(SIP/1002)
 

> Running as user 'asterisk' Running
  Running under group 'asterisk' Connected to
 to  Asterisk 11.5.0 currently running on
 on  pbx (pid = 17967)  [Aug 11 17:27:20]
  ERROR[17985][C-00000003]:
  rtpengine.c:259 astrtpinstancenew:
  17:27:20] ERROR[17985][C-00000003]: > rtp_engine.c:259 ast_rtp_instance_new: > No RTP engine was found. Do you have
  have > one loaded?  [Aug 11 17:27:20]
  NOTICE[17985][C-00000003]:
  chansip.c:25282
  handlerequest_invite: 17:27:20] NOTICE[17985][C-00000003]: > chan_sip.c:25282 > handle_request_invite: Failed to
  to > authenticate device "1002"
  <sip:1002@192.168.0.100>;tag=1665125730
  pbx*CLI>