Пытаюсь настроить Asterisk.
Вот мои конфиги:
sip.conf
Код: Выделить всё
[general]
context=default
allowguest=no
allowoverlap=no
udpbindaddr=192.168.111.111
tcpenable=no
tcpbindaddr=192.168.111.111
srvlookup=yes
language=ru
callevents=yes
alwaysauthreject = yes
t38pt_udptl=no
[asterisk-1]
type=friend
secret=qwerty
nat=no
host=192.168.100.111
dtmfmode=rfc2833
context=to-asterisk1
disallow=all
canreinvite=no
insecure=invite,port
allow=alaw
qualify=yes
[codecs](!)
disallow=all
allow=alaw
[phones](!,codecs)
type=peer
host=dynamic
canreinvite=no
context=from-phones
subscribecontext=test-phones
callcounter=yes
secret=test123
nat=no
[localip](!,phones)
qualify=yes
[201](localip)
[202](localip)
[203](localip)
[204](localip)
[205](localip)
[206](localip)
[207](localip)
[208](localip)
[209](localip)
[210](localip)
extensions.conf
Код: Выделить всё
[general]
[default]
exten => _X.,1,Hangup
[all_phones]
exten => _[12345]XX,1,Dial(SIP/asterisk-1/${EXTEN})
same => n,Hangup
exten => _98XXXXXXXXXX,1,Dial(SIP/asterisk-1/${EXTEN})
same => n,Hangup
[test-phones]
exten => 201,1,Dial(SIP/201,90,t)
same => n,Hangup
exten => 202,1,Dial(SIP/202,90,t)
same => n,Hangup
exten => 203,1,Dial(SIP/203,90,t)
same => n,Hangup
exten => 204,1,Dial(SIP/204,90,t)
[from-phones]
include => test-phones
include => all_phones
[to-asterisk1]
include => test-phones
exten => 201,1,Dial(SIP/201,90,t)
same => n,Hangup
exten => 202,1,Dial(SIP/202,90,t)
same => n,Hangup
exten => 203,1,Dial(SIP/203,90,t)
same => n,Hangup
exten => 204,1,Dial(SIP/204,90,t)
same => n,Hangup
exten => _[12345]XX,1,Dial(SIP/asterisk-1/${EXTEN})
same => n,Hangup
exten => _98XXXXXXXXXX,1,Dial(SIP/asterisk-1/${EXTEN})
same => n,Hangup
Код: Выделить всё
Name/username Host Dyn Forcerport ACL Port Status
201/201 192.168.0.101 D 5060 OK (4 ms)
202/202 192.168.0.102 D 5060 OK (5 ms)
203 (Unspecified) D 0 UNKNOWN
204 (Unspecified) D 0 UNKNOWN
205 (Unspecified) D 0 UNKNOWN
206 (Unspecified) D 0 UNKNOWN
207 (Unspecified) D 0 UNKNOWN
208 (Unspecified) D 0 UNKNOWN
209 (Unspecified) D 0 UNKNOWN
210 (Unspecified) D 0 UNKNOWN
asterisk-1 192.168.100.111 5060 OK (1 ms)
sip show peer 202
Код: Выделить всё
* Name : 202
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-phones
Subscr.Cont. : test-phones
Language : ru
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 36
Insecure : no
Force rport : No
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.0.102:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 498
SIP Options : (none)
Codecs : 0x8 (alaw)
Codec Order : (alaw:20)
Auto-Framing : No
Status : OK (5 ms)
Useragent : Voip Phone 1.0
Reg. Contact : sip:202@192.168.0.102:5060
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
А при звонке с телефона 202 на номер 333 звонок не проходит выдаётся ошибка:
[Mar 12 09:02:05] NOTICE[3124]: chan_sip.c:20870 handle_response_invite: Failed to authenticate on INVITE to '"Test202" <sip:202@192.168.111.111>;tag=as6bdc9894'
Оба телефона Fanvil BW210, настройки телефонов абсолютно одинаковые.
