Dear All, I have the below settings on my asterisk server and I need to know if there is a any problem in a setting regarding performance or security..Please check and let me know:
Global Settings: ---------------- SIP Port: 5060 Bindaddress: IP_ADDRESS Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: Yes Promsic. redir: No SIP domain support: Yes 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: No 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: No RFC2833 Compensation: No SIP realtime: Enabled Global Signalling Settings: --------------------------- Codecs: 0x10e (gsm|ulaw|alaw|g729) Codec Order: g729:20,ulaw:20,alaw:20,gsm:20 T1 minimum: 100 Relax DTMF: 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: 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 localhost*CLI> 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 Realtime SIP Settings: ---------------------- Realtime Peers: Yes Realtime Users: Yes Cache Friends: No Update: Yes Ignore Reg. Expire: No Save sys. name: No Auto Clear: 120 Thanks a lot
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