Asterisk SVN-trunk-r102700, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= NOTE: This is a development version of Asterisk, and should not be used in production installations. Connected to Asterisk SVN-trunk-r102700 currently running on laptop-asterisk (pid = 6054) laptop-asterisk*CLI> sip show settings laptop-asterisk*CLI> Global Settings: laptop-asterisk*CLI> ---------------- laptop-asterisk*CLI> SIP Port: 5060 laptop-asterisk*CLI> Bindaddress: 0.0.0.0 laptop-asterisk*CLI> Videosupport: No laptop-asterisk*CLI> Textsupport: No laptop-asterisk*CLI> AutoCreatePeer: No laptop-asterisk*CLI> MatchAuthUsername: No laptop-asterisk*CLI> Allow unknown access: Yes laptop-asterisk*CLI> Allow subscriptions: Yes laptop-asterisk*CLI> Enable call counters: No laptop-asterisk*CLI> Allow overlap dialing: Yes laptop-asterisk*CLI> Promsic. redir: No laptop-asterisk*CLI> SIP domain support: No laptop-asterisk*CLI> Call to non-local dom.: Yes laptop-asterisk*CLI> URI user is phone no: No laptop-asterisk*CLI> Our auth realm asterisk laptop-asterisk*CLI> Realm. auth: No laptop-asterisk*CLI> Always auth rejects: No laptop-asterisk*CLI> Call limit peers only: No laptop-asterisk*CLI> Direct RTP setup: No laptop-asterisk*CLI> User Agent: Asterisk PBX SVN-trunk-r102700 laptop-asterisk*CLI> SDP Session Name: Asterisk PBX SVN-trunk-r102700 laptop-asterisk*CLI> SDP Owner Name: root laptop-asterisk*CLI> Reg. context: (not set) laptop-asterisk*CLI> Regexten on Qualify: No laptop-asterisk*CLI> Caller ID: Unknown laptop-asterisk*CLI> From: Domain: laptop-asterisk*CLI> Record SIP history: Off laptop-asterisk*CLI> Call Events: Off laptop-asterisk*CLI> IP ToS SIP: CS0 laptop-asterisk*CLI> IP ToS RTP audio: CS0 laptop-asterisk*CLI> IP ToS RTP video: CS0 laptop-asterisk*CLI> IP ToS RTP text: CS0 laptop-asterisk*CLI> 802.1p CoS SIP: 4 laptop-asterisk*CLI> 802.1p CoS RTP audio: 5 laptop-asterisk*CLI> 802.1p CoS RTP video: 6 laptop-asterisk*CLI> 802.1p CoS RTP text: 5 laptop-asterisk*CLI> T38 fax pt UDPTL: No laptop-asterisk*CLI> RFC2833 Compensation: No laptop-asterisk*CLI> Jitterbuffer enabled: No laptop-asterisk*CLI> Jitterbuffer forced: No laptop-asterisk*CLI> Jitterbuffer max size: -1 laptop-asterisk*CLI> Jitterbuffer resync: -1 laptop-asterisk*CLI> Jitterbuffer impl: laptop-asterisk*CLI> Jitterbuffer log: No laptop-asterisk*CLI> SIP realtime: Disabled laptop-asterisk*CLI> Qualify Freq : 60000 ms laptop-asterisk*CLI> Network Settings: laptop-asterisk*CLI> --------------------------- laptop-asterisk*CLI> SIP address remapping: Disabled, no localnet list laptop-asterisk*CLI> Externhost: laptop-asterisk*CLI> Externip: 0.0.0.0:0 laptop-asterisk*CLI> Externrefresh: 10 laptop-asterisk*CLI> Internal IP: 127.0.1.1:5060 laptop-asterisk*CLI> STUN server: 0.0.0.0:0 laptop-asterisk*CLI> Global Signalling Settings: laptop-asterisk*CLI> --------------------------- laptop-asterisk*CLI> Codecs: 0xc (ulaw|alaw) laptop-asterisk*CLI> Codec Order: ulaw:20,alaw:20 laptop-asterisk*CLI> Relax DTMF: No laptop-asterisk*CLI> Compact SIP headers: No laptop-asterisk*CLI> RTP Keepalive: 0 (Disabled) laptop-asterisk*CLI> RTP Timeout: 0 (Disabled) laptop-asterisk*CLI> RTP Hold Timeout: 0 (Disabled) laptop-asterisk*CLI> MWI NOTIFY mime type: application/simple-message-summary laptop-asterisk*CLI> DNS SRV lookup: Yes laptop-asterisk*CLI> Pedantic SIP support: No laptop-asterisk*CLI> Reg. min duration 60 secs laptop-asterisk*CLI> Reg. max duration: 3600 secs laptop-asterisk*CLI> Reg. default duration: 120 secs laptop-asterisk*CLI> Outbound reg. timeout: 20 secs laptop-asterisk*CLI> Outbound reg. attempts: 0 laptop-asterisk*CLI> Notify ringing state: Yes laptop-asterisk*CLI> Notify hold state: No laptop-asterisk*CLI> SIP Transfer mode: open laptop-asterisk*CLI> Max Call Bitrate: 384 kbps laptop-asterisk*CLI> Auto-Framing: No laptop-asterisk*CLI> Outb. proxy: laptop-asterisk*CLI> Session Timers: Accept laptop-asterisk*CLI> Session Refresher: uas laptop-asterisk*CLI> Session Expires: 1800 secs laptop-asterisk*CLI> Session Min-SE: 90 secs laptop-asterisk*CLI> Timer T1: 500 laptop-asterisk*CLI> Timer T1 minimum: 100 laptop-asterisk*CLI> Timer B: 32000 laptop-asterisk*CLI> Default Settings: laptop-asterisk*CLI> ----------------- laptop-asterisk*CLI> Context: incoming-from-sip laptop-asterisk*CLI> Nat: RFC3581 laptop-asterisk*CLI> DTMF: inband laptop-asterisk*CLI> Qualify: 0 laptop-asterisk*CLI> Use ClientCode: No laptop-asterisk*CLI> Progress inband: Never laptop-asterisk*CLI> Language: se laptop-asterisk*CLI> MOH Interpret: default laptop-asterisk*CLI> MOH Suggest: laptop-asterisk*CLI> Voice Mail Extension: asterisk laptop-asterisk*CLI> ---- laptop-asterisk*CLI> exit