Elastix за OpenVPN

[general]
tos_sip=cs3
tos_audio=ef
tos_video=af41
rtcachefriends=yes
callcounter=yes
alwaysauthreject=yes
maxexpiry=3600
minexpiry=60
defaultexpiry=120
qualifyfreq=60
qualifygap=100
registertimeout=20
registerattempts=0
rtptimeout=60
rtpholdtimeout=300
videosupport=yes
maxcallbitrate=384
faxdetect=yes
t38pt_udptl=yes
directmedia=no
notifyringing=yes
notifyhold=yes
dtmfmode=auto
relaxdtmf=yes
trustrpid=yes
sendrpid=no
useragent=Elastix 3.0
vmexten=*97
language=en
disallow=all
allow=gsm
allow=ulaw
allow=alaw
g726nonstandard=en
accept_outofcall_message=yes
auth_message_requests=yes
bindaddr=0.0.0.0
bindport=5060
outboundproxy=127.0.0.1
outboundproxyport=5060
localnet=192.168.0.0/255.255.255.0
localnet=192.168.1.0/255.255.255.0
nat=force_rport,comedia


#include sip_general_custom.conf
#include sip_register.conf
#include sip_custom.conf

Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: Yes
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: Yes
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: Elastix 3.0
SDP Session Name: Asterisk PBX 11.13.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: Yes
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: Yes
T.38 EC mode: FEC
T.38 MaxDtgrm: 4294967295
SIP realtime: Enabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
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
Externhost:
Externaddr: (null)
Externrefresh: 10
Localnet: 192.168.0.0/255.255.255.0
192.168.0.0/255.255.255.0
192.168.1.0/255.255.255.0

Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw)
Codec Order: gsm:20,ulaw:20,alaw:20
Relax DTMF: Yes
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 60
RTP Hold Timeout: 300
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
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: 127.0.0.1
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: Yes
DTMF: auto
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language: en
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

Realtime SIP Settings:
----------------------
Realtime Peers: Yes
Realtime Regs: No
Cache Friends: Yes
Update: Yes
Ignore Reg. Expire: No
Save sys. name: No
Auto Clear: 120 (Disabled)

----