Имеется Asterisk 13 и AddPac gs1002. Входящие работают нормально, но с исходящими беда
Using SIP RTP CoS mark 5 -- Executing [89679328252@_sip:1] Goto("SIP/100-00000004", "dial_to_beeline,89679328252,1),NoOP(Исходящий на номер 89679328252 с номера 100") in new stack -- Goto (dial_to_beeline,89679328252,1) -- Executing [89679328252@dial_to_beeline:1] NoOp("SIP/100-00000004", ""Звоним на билайновский номер 89679328252 с номера 100"") in new stack -- Executing [89679328252@dial_to_beeline:2] Set("SIP/100-00000004", "CALLERID(number)=79034814792") in new stack -- Executing [89679328252@dial_to_beeline:3] Dial("SIP/100-00000004", "SIP/79034814792/89679328252,30,m") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/79034814792/89679328252 -- Started music on hold, class 'default', on channel 'SIP/100-00000004' > 0xb6843868 -- Probation passed - setting RTP source address to 192.168.1.6:52890 == Everyone is busy/congested at this time (1:0/0/1) -- Stopped music on hold on SIP/100-00000004 -- Auto fallthrough, channel 'SIP/100-00000004' status is 'CHANUNAVAIL'
Конфиг AddPac:
!
! APOS(tm) configuration saved from vty
! 2016/10/17 19:58:47
!
version 8.51.011
!
hostname GS1002
!
username root password ********* administrator
!
!
script ntpdate default
server ip time.nist.gov
server ip time.windows.com
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 192.168.1.100 255.255.255.0
speed auto
no qos-control
!
interface FastEthernet0/1
ip address 192.168.10.1 255.255.255.0
speed auto
no qos-control
!
!
!
!
http server
!
logging command
logging event 4-warning
logging on
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
protocol sip
dtmf-relay rfc-2833
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
no call-barring unconfigured-ip-address
no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
connection plar 7929******
ring detect-timeout 70
dial-tone-generate
caller-id enable
caller-id type etsi
!
!
! GSM
voice-port 0/1
connection plar 7903*******
ring detect-timeout 70
dial-tone-generate
caller-id enable
caller-id type etsi
!
!
! FXO
voice-port 0/2
no caller-id enable
!
!
! FXO
voice-port 0/3
no caller-id enable
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 00T
port 0/0
call-waiting
user-name 7929881*****
user-password ***********
translate-outgoing called-number 0
preference 1
diversion 1
!
dial-peer voice 1 pots
destination-pattern 01T
port 0/1
call-waiting
user-name 79034******
user-password *********
translate-outgoing called-number 1
preference 2
diversion 2
!
!
!
! Voip peer configuration.
!
dial-peer voice 2000 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 1
no vad
dtmf-relay rtp-2833
description asterisk
!
!
!
dial-peer call-hold h
dial-peer call-transfer h
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.1.8
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
!
!
!
! Translation Rule configuration.
!
translation-rule 0
rule 0 007T 8T
!
translation-rule 1
rule 0 017T 8T
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-username addpac
sip-password *********
sip-server 192.168.1.8 5060 1
timeout treg 400
called-party-number to-field
remote-party-id
session-refresh update
register e164
!
!
! Tones
!
!
! SMS delivery configuration
!
sms-delivery
!
!
!
!
voip-interface ip FastEthernet0/0
!
line console
!
line vty
!
mobile dev-restart-by-unreg 180
mobile failed-call-retry 0
mobile ussd inter-frame-gap 100
mobile ussd balance-interval 120
mobile ussd retry-count 2
mobile ussd retry-interval 5
mobile ussd response-protection-time 5
mobile dev-restart-by-unknown-error
mobile cell-monitor 30
!
mobile 0/0
gsm sms-language utf8
!
mobile 0/1
gsm sms-language utf8
!
sip.conf
[general]
trustrpid=yes
tcpenable=yes
useragent=D-link ; типа реальное железо. чтобы меньше привлекать внимание хацкеров которые ищут астериски
;externip=176.122.61.94
localnet=192.168.1.0/24
qualify=yes ; проверка доступности абонента - 2s - если больше, что считаем что недоступен
prematuremedia = no
progressinband = never
srvlookup=no
canreinvite=no ; разрешает (yes) или запрещает (no) установку прямого соединения(минуя Asterisk).
directmedia=no ; гнать трафик напрямую
allowguest = no ; запрет регистрации "левых" аккунтов
transfer=yes ; запрет трансфера вызовов глобально, включать вручную для нужных пиров
allowsubscribe=no ; отказ от использования voicemail и соответствующего спама в консоли
alwaysauthreject=yes ; на REGISTER Asterisk станет отвечать «401 Unathorized»
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a SIP
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize=300
jbimpl = adaptive ; Jitterbuffer implementation, used on the receiving side of a SIP
jbresyncthreshold=100 ;adaptive jbimpl had troubles without thresh... Asterisk 1.6.2.9.
;jblog = yes ; Enables jitterbuffer frame logging. Defaults to "no".
context=default ; всех левых в дефротный контекст на отбой
relaxdtmf=yes
dtmfmode=auto
disallow=all
;allow=g729
;allow=g723
allow=alaw
allow=ulaw
allow=gsm
bindport=5060
[addpac_channels](!) ; шаблон дублирующихся настроек для каналов шлюза
host=dynamic
;deny=0.0.0.0/0
permit=192.168.1.100
fromdomain=192.168.1.100
type=friend
context=from_trunk ; входящие с SIP попадают в этот контекст в extensions.conf
qualify=yes
nat=force_rport,comedia
canreinvite=no
insecure=port,invite ; игнорировать порт и инвайт
disallow=all
allow=alaw
allow=ulaw
allow=gsm
maxcallbitrate=64
dtmfmode=rfc2833
port=5060
;directmedia=no
[7903********](addpac_channels)
defaultuser=********
secret=********
call-limit=2
callerid=79034********
relaxdtmf=yes
[79298********](addpac_channels)
defaultuser=********
secret=********
call-limit=1
callerid=********
relaxdtmf=yes
exstensions.conf
]
[general]
static=yes
writeprotect=yes
autofallthrough=yes ; завершить вызов после всех команд екстеншена
clearglobalvars=no
;priorityjumping=no
[_sip]
exten => _+7.,1,Goto(_sip,${EXTEN:1:11},1) ; обрезание плюса в начале номера
exten => _[7,8]9XXXXXXXXX,1,Goto(dial_to_beeline,${EXTEN},1),NoOP(Исходящий на номер ${EXTEN} с номера ${CALLERID(num)})
[dial_to_beeline] ; контекст исходящих звонков на билайн
exten => _[7,8]9.,1,NoOP("Звоним на билайновский номер ${EXTEN} с номера ${CALLERID(num)}")
same => n,Set(CALLERID(number)=790348****)
same => n,Dial(SIP/7903481****/${EXTEN},30,m)
Дебаг addpac
: ****** Call Created status(InitiatedByNet) ver(8.51:2011-02-06-00-00) time(1476739401) ****
2 : Receive INVITE Request
3 : Found inbound voip peer(2000) result(2) peer->fixedPatternSize(0) mostMatchingSize(-1)
4 : Found inbound voip peer by dest-pattern id(2000)
5 : From Net - calledParty(79679328252) callingParty(79034814792)
6 : Unmatched
7 : Terminated from(fffffff7) this(Local:InvalidNumber) before(NULL) forced(0) time(1476739401)
8 : Call FROM <79034814792> terminated reason(Local:InvalidNumber)
9 : Receive ACK Request
10 : Set Terminated Success for 102 INVITE
В чем может быть проблема?
Простой
