rendered paste bodyGlobal Settings:
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SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
Always auth rejects: No
Call limit peers only: Yes
Direct RTP setup: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS SIP: none
IP ToS RTP audio: none
IP ToS RTP video: none
T38 fax pt UDPTL: Yes
RFC2833 Compensation: No
SIP realtime: Disabled
Global Signalling Settings:
---------------------------
Codecs: 0x104 (ulaw|g729)
Codec Order: g729:20,ulaw:20
T1 minimum: 100
Relax DTMF: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 300
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Default Settings:
-----------------
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk