Попробовал по ссылке вносить настройки в conf файлы, в сеть выходит, но не звонит. Пишет "не найдено".
Попробовал установить систему заново по инструкции по ссылке:
http://voxlink.ru/kb/asterisk-installation/from-source/
Все работает по UDP и регистрация и звонок.
Работает только со следующими конфигурационными файлами:
sip.conf
Код: Выделить всё
[general]
;глобальные значения переменных канала SIP
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context=default
[301]
type=peer
secret=***
qualify=yes
pickupgroup=1
nat=yes
mailbox=301@device
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=gy29
dial=SIP/301
context=from-internal
canreinvite=no
callgroup=1
callerid=Alexey <301>
call-limit=2
[302]
type=peer
secret=***
qualify=yes
pickupgroup=1
nat=yes
mailbox=302@device
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=gy29
dial=SIP/302
context=from-internal
canreinvite=no
callgroup=1
callerid=Alexey <302>
call-limit=2
Код: Выделить всё
[default]
[from-internal]
exten => 301,1,Dial(SIP/301,30,t)
exten => 302,1,Dial(SIP/302,30,t)
Я не помню где я взял такую конфигурацию extensions.conf, но по другим инструкциям регистрируется в сети, но не звонит.
Код: Выделить всё
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 11.0.0
SDP Session Name: Asterisk PBX 11.0.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw)
Codec Order: gsm:20,ulaw:20,alaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Record on feature: automon
Record off feature: automon
Force rport: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
Если сейчас начну настраивать TLS+SRTP и пересоберу asterisk, но не будит регистрироваться в сети, не каких ошибок и логов.
