Проблема такая: есть один сервер и два оффиса, два номера и два контекста соответственно
При звонке на любой из номеров вызывается контекст какогото одного оператора, и в одном случае это соответствует номеру, в другом - не соответствует.куда копать даже не знаю, подскажите
Вот лог консоли:
[root@i4 asterisk]# asterisk -r
Setting max files open to 1000
Verbosity is at least 3
isp4*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
qwerty.cnt.ru:5060 N 8499502XXXX 345 Registered Tue, 22 Mar 2011 16:31:17
qwerty.cnt.ru:5060 N 8499502YYYY 345 Registered Tue, 22 Mar 2011 16:31:17
2 SIP registrations.
#############################
#Зовнок на номер 8499502XXXX#
#############################
[2011-03-22 16:37:01] == Using SIP RTP CoS mark 5
[2011-03-22 16:37:01] -- Executing [000@abay-office:1] Answer("SIP/abay-centel-00000004", "") in new stack
[2011-03-22 16:37:01] -- Executing [000@abay-office:2] Playback("SIP/abay-centel-00000004", "abay01") in new stack
[2011-03-22 16:37:01] -- <SIP/abay-centel-00000004> Playing 'abay01.ulaw' (language 'ru')
[2011-03-22 16:37:02] == Spawn extension (abay-office, 000, 2) exited non-zero on 'SIP/abay-centel-00000004'
#############################
#Зовнок на номер 8499502YYYY#
#############################
[2011-03-22 16:38:32] == Using SIP RTP CoS mark 5
[2011-03-22 16:38:32] -- Executing [000@abay-office:1] Answer("SIP/abay-centel-00000005", "") in new stack
[2011-03-22 16:38:32] -- Executing [000@abay-office:2] Playback("SIP/abay-centel-00000005", "abay01") in new stack
[2011-03-22 16:38:32] -- <SIP/abay-centel-00000005> Playing 'abay01.ulaw' (language 'ru')
[2011-03-22 16:38:34] == Spawn extension (abay-office, 000, 2) exited non-zero on 'SIP/abay-centel-00000005'
конфиги:
[root@i4 asterisk]# cat sip.conf
[general]
context=default ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
match_auth_username=yes ; if available, match user entry using the
allowoverlap=yes ; Disable overlap dialing support. (Default is yes)
allowtransfer=yes ; Disable all transfers (unless enabled in peers or users)
udpbindaddr=193.9.17.85 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no ; Enable server for incoming TCP connections (default is no)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
disallow=all ; First disallow all codecs
allow=alaw
allow=ulaw
allow=g729
language=ru ; Default language setting for all users/peers
usereqphone = yes ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
videosupport=yes ; Turn on support for SIP video. You need to turn this
maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=no ; generate manager events when sip ua
auth_options_requests = no ; Enabling this option will authenticate OPTIONS requests just like
;contactdeny=0.0.0.0/0.0.0.0
;contactpermit=109.73.4.0/255.255.255.0
shrinkcallerid=yes ; on by default
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
; (default is off - zero)
stunaddr = stun.sipnet.ru:3478
externrefresh = 60
[abay-centel]
outboundproxy=213.85.168.52
context=abay-office
secret = xSbwJTdV
defaultuser = 8499502YYYY
trunkname = abay-centel
hasexten = no
hassip = yes
hasiax = no
host = qwerty.cnt.ru
insecure = invite
;fromuser = 8499502YYYY
fromdomain = qwerty.cnt.ru
type = peer
callbackextension = 000
disallow = all
allow = alaw
allow = ulaw
allow = g729
nat = no
canreinvite = nonat
dtmfmode = rfc2833
[amki-centel]
outboundproxy=213.85.168.52
context=amki-office
secret = 5DMVS4Dk
defaultuser = 8499502XXXX
trunkname = amki-centel
hasexten = no
hassip = yes
hasiax = no
host = qwerty.cnt.ru
insecure = invite
fromuser = 8499502XXXX
fromdomain = qwerty.cnt.ru
type = peer
callbackextension = 000
disallow = all
allow = alaw
allow = ulaw
allow = g729
nat = no
canreinvite = nonat
dtmfmode = rfc2833
[root@i4 asterisk]# cat extensions.conf
[default]
exten => 000,1,Hangup
[amki-office]
exten => _8XXXXXXXXXX,1,set(CALLERID(all)=8499502XXXX)
exten => _8XXXXXXXXXX,n,Dial(SIP/amki-centel/${EXTEN},120)
exten => 000,1,Answer()
exten => 000,n,Set(fname=${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-${CALLERID(number)}-${EXTEN})
exten => 000,n,MixMonitor(/var/lib/asterisk/record/in_amki_${fname}.wav)
exten => 000,1,Dial(IAX2/amki-manager,rt)
exten => 000,n,Hangup
[abay-office]
exten => _8XXXXXXXXXX,1,set(CALLERID(all)=8499502YYYY)
exten => _8XXXXXXXXXX,n,Dial(SIP/abay-centel/${EXTEN},120)
;start
exten => 000,1,Answer()
exten => 000,n,Playback(abay01) ; приветствует звонящего
exten => 000,n,set(DigitTimeout=5) ; время ожидания нажатия
exten => 000,n,set(ResponseTimeout=2) ; время ожидания ответа
exten => 000,n,WaitExten() ; ждем 5 сек ввода добавочного номера
exten => 000,n,GotoIfTime(16:25-9:00|sat|*|*?abay-office,000,100) ; рабочие часы в субботу с 11 до 16.30
exten => 000,n,GotoIfTime(17:50-11:00|*|*|*?abay-office,000,100) ; в остальные дни с 11 до 18
exten => 000,n,Playback(abay02) ; вы переводитесь на оператора, разговор записывается
exten => 000,n,Set(fname=${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-${CALLERID(number)}-${EXTEN})
exten => 000,n,MixMonitor(/var/lib/asterisk/record/in_abay_${fname}.wav)
exten => 000,n,Queue(abay) ; включаем микшер и переводим разговор в очередь
exten => 000,n,Hangup
ось- центось
Asterisk 1.8.3.2
Вариант для распечатки
