Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
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.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux8 (pid = 6260)
Verbosity was 4 and is now 10
linux8*CLI>
linux8*CLI>
linux8*CLI> sip show
channel channels channelstats domains history inuse mwi objects peer peers registry sched settings subscriptions tcp users
user
linux8*CLI> sip show users
Username Secret Accountcode Def.Context ACL NAT
2402 1234 default No RFC3581
2401 1234 default No RFC3581
linux8*CLI> sip show ch
channels channelstats channel
linux8*CLI> sip show channel
Usage: sip show channel <call-id>
Provides detailed status on a given SIP dialog (identified by SIP call-id).
linux8*CLI> sip show mwi
Host Username Mailbox Subscribed
linux8*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
linux8*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
linux8*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
linux8*CLI> sip show pee
peers peer
linux8*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2401/2401 192.168.80.201 D 5060 Unmonitored
2402/2402 192.168.80.205 D 5618 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux8*CLI> sip show peer
Usage: sip show peer <name> [load]
Shows all details on one SIP peer and the current status.
Option "load" forces lookup of peer in realtime storage.
linux8*CLI> sip show peer
2402 2401
linux8*CLI> sip show peer 2401
* Name : 2401
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 464
Insecure : no
Nat : RFC3581
ACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.80.201 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 2401
SIP Options : (none)
Codecs : 0x8000e (gsm|ulaw|alaw|h263)
Codec Order : (none)
Auto-Framing : No
100 on REG : No
Status : Unmonitored
Useragent : Cisco-CP7960G/7.5
Reg. Contact : sip:2401@192.168.80.201:5060
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
Parkinglot :
linux8*CLI> sip show peer 2402
* Name : 2402
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 3321
Insecure : no
Nat : RFC3581
ACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.80.205 Port 5618
Defaddr->IP : 0.0.0.0 Port 5060
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 2402
SIP Options : (none)
Codecs : 0x8000e (gsm|ulaw|alaw|h263)
Codec Order : (none)
Auto-Framing : No
100 on REG : No
Status : Unmonitored
Useragent : X-Lite 4 release 4.1 stamp 63214
Reg. Contact : sip:2402@192.168.80.205:5618;rinstance=d8f217608b7fa924
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
Parkinglot :
linux8*CLI> sip show
No such command 'sip show' (type 'core show help sip show' for other possible commands)
linux8*CLI> sip show
channel channels channelstats domains history inuse mwi objects peer peers registry sched settings subscriptions tcp users
user
linux8*CLI> sip show objects
-= Peer objects: 2 static, 0 realtime, 0 autocreate =-
name: 2402
type: peer
objflags: 0
refcount: 6
name: 2401
type: peer
objflags: 0
refcount: 3
-= Registry objects: 0 =-
-= Dialog objects:
linux8*CLI> sip show settings \
Global Settings:
----------------
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: No
Direct RTP setup: No
User Agent: Asterisk PBX 1.6.2.9-2+squeeze3
SDP Session Name: Asterisk PBX 1.6.2.9-2+squeeze3
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
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
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
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externip: 0.0.0.0:0
Externrefresh: 10
Internal IP: 192.168.15.28:5060
STUN server: 0.0.0.0:0
Global Signalling Settings:
---------------------------
Codecs: 0x8000e (gsm|ulaw|alaw|h263)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: 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
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
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
----
linux8*CLI> sip show settings
Global Settings:
----------------
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: No
Direct RTP setup: No
User Agent: Asterisk PBX 1.6.2.9-2+squeeze3
SDP Session Name: Asterisk PBX 1.6.2.9-2+squeeze3
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
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
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
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externip: 0.0.0.0:0
Externrefresh: 10
Internal IP: 192.168.15.28:5060
STUN server: 0.0.0.0:0
Global Signalling Settings:
---------------------------
Codecs: 0x8000e (gsm|ulaw|alaw|h263)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: 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
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
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
----
linux8*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2401/2401 192.168.80.201 D 5060 Unmonitored
2402/2402 192.168.80.205 D 5618 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux8*CLI>
linux8*CLI>
linux8*CLI>
linux8*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2401/2401 192.168.80.201 D 5060 Unmonitored
2402/2402 192.168.80.205 D 5618 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux8*CLI> sip show
channel channels channelstats domains history inuse mwi objects peer peers registry sched settings subscriptions tcp users
user
linux8*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
linux8*CLI> sip show settings
Global Settings:
----------------
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: No
Direct RTP setup: No
User Agent: Asterisk PBX 1.6.2.9-2+squeeze3
SDP Session Name: Asterisk PBX 1.6.2.9-2+squeeze3
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
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
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
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externip: 0.0.0.0:0
Externrefresh: 10
Internal IP: 192.168.15.28:5060
STUN server: 0.0.0.0:0
Global Signalling Settings:
---------------------------
Codecs: 0x8000e (gsm|ulaw|alaw|h263)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: 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
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
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
----
linux8*CLI> quit
Executing last minute cleanups