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История изменений [назад]

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ответил 2012-06-13 10:19:39 +0400

a.r.t. Gravatar a.r.t.

sip show settings

ip ro show 1x.6x.18.0/30 dev eth1 proto kernel scope link src 10.60.18.2

4x.4x.15x.8x/29 dev eth2 proto kernel scope link src 46.46.153.90

192.168.0.0/22 dev eth0 proto kernel scope link src 192.168.0.38

169.254.0.0/16 dev eth2 scope link

default via 1x.60.18.1 dev eth1

default via 4x.4x.153.89 dev eth2

sip show settings

ip ro show show

1x.6x.18.0/30 dev eth1 proto kernel scope link src 10.60.18.2

4x.4x.15x.8x/29 dev eth2 proto kernel scope link src 46.46.153.90

192.168.0.0/22 dev eth0 proto kernel scope link src 192.168.0.38

169.254.0.0/16 dev eth2 scope link

default via 1x.60.18.1 dev eth1

default via 4x.4x.153.89 dev eth2

sip asterisk -rx "sip show settings

settings"


UDP SIP Port: 5060

UDP Bindaddress: 0.0.0.0

TCP SIP Port: 5060

TCP Bindaddress: 0.0.0.0

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: Yes

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: Yes

Direct RTP setup: No

User Agent: FPBX-2.5.2(1.6.2.6)

SDP Session Name: Asterisk PBX 1.6.2.6

SDP Owner Name: root

Reg. context: (not set)

Regexten on Qualify: No

Caller ID: Unknown

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: 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 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: 127.0.0.1:5060 STUN server: 0.0.0.0:0

Global Signalling Settings:

Codecs: 0xc (ulaw|alaw) Codec Order: ulaw:20,alaw:20 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: Yes 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: TCP,UDP Outbound transport: UDP Context: from-sip-external Nat: RFC3581 DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: MOH Interpret: default MOH Suggest: Voice Mail Extension: *97

ip ro show

1x.6x.18.0/30 dev eth1 proto kernel scope link src 10.60.18.2

4x.4x.15x.8x/29 dev eth2 proto kernel scope link src 46.46.153.90

192.168.0.0/22 dev eth0 proto kernel scope link src 192.168.0.38

169.254.0.0/16 dev eth2 scope link

default via 1x.60.18.1 dev eth1

default via 4x.4x.153.89 dev eth2

asterisk -rx "sip show settings"


UDP SIP Port: 5060

UDP Bindaddress: 0.0.0.0

TCP SIP Port: 5060

TCP Bindaddress: 0.0.0.0

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: Yes

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: Yes

Direct RTP setup: No

User Agent: FPBX-2.5.2(1.6.2.6)

SDP Session Name: Asterisk PBX 1.6.2.6

SDP Owner Name: root

Reg. context: (not set)

Regexten on Qualify: No

Caller ID: Unknown

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: CS3 CS3

IP ToS RTP audio: EF EF

IP ToS RTP video: AF41 AF41

IP ToS RTP text: CS0 CS0

802.1p CoS SIP: 4 4

802.1p CoS RTP audio: 5 5

802.1p CoS RTP video: 6 6

802.1p CoS RTP text: 5 5

Jitterbuffer enabled: No No

Jitterbuffer forced: No No

Jitterbuffer max size: -1 -1

Jitterbuffer resync: -1 -1

Jitterbuffer impl: impl:

Jitterbuffer log: No

Network Settings:

SIP address remapping: Disabled, no localnet list list

Externhost: <none> <none>

Externip: 0.0.0.0:0 0.0.0.0:0

Externrefresh: 10 10

Internal IP: 127.0.0.1:5060 127.0.0.1:5060

STUN server: 0.0.0.0:0

Global Signalling Settings:

Codecs: 0xc (ulaw|alaw) (ulaw|alaw)

Codec Order: ulaw:20,alaw:20 ulaw:20,alaw:20

Relax DTMF: No No

RFC2833 Compensation: No No

Compact SIP headers: No No

RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) (Disabled)

MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Yes

Pedantic SIP support: No No

Reg. min duration 60 secs secs

Reg. max duration: 3600 secs secs

Reg. default duration: 120 secs secs

Outbound reg. timeout: 20 secs secs

Outbound reg. attempts: 0 0

Notify ringing state: Yes Yes

Include CID:          No
 

Notify hold state: Yes Yes

SIP Transfer mode: open open

Max Call Bitrate: 384 kbps kbps

Auto-Framing: No No

Outb. proxy: <not set=""> set="">

Session Timers: Accept Accept

Session Refresher: uas uas

Session Expires: 1800 secs secs

Session Min-SE: 90 secs secs

Timer T1: 500 500

Timer T1 minimum: 100 100

Timer B: 32000 32000

No premature media: Yes

Default Settings:

Allowed transports: TCP,UDP TCP,UDP

Outbound transport: UDP UDP

Context: from-sip-external from-sip-external

Nat: RFC3581 RFC3581

DTMF: rfc2833 rfc2833

Qualify: 0 0

Use ClientCode: No No

Progress inband: Never Never

Language: MOH Interpret: default default

MOH Suggest: Voice Mail Extension: *97

ip ro show

1x.6x.18.0/30 dev eth1 proto kernel scope link src 10.60.18.2

10.60.18.2ist item
4x.4x.15x.8x/29 dev eth2 proto kernel scope link src 46.46.153.90


192.168.0.0/22 dev eth0 proto kernel scope link src 192.168.0.38


169.254.0.0/16 dev eth2 scope link


default via 1x.60.18.1 dev eth1


default via 4x.4x.153.89 dev eth2

asterisk -rx "sip show settings"



UDP SIP Port: 5060


UDP Bindaddress: 0.0.0.0


TCP SIP Port: 5060


TCP Bindaddress: 0.0.0.0


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: Yes


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: Yes


Direct RTP setup: No


User Agent: FPBX-2.5.2(1.6.2.6)


SDP Session Name: Asterisk PBX 1.6.2.6

1.6.2.6
SDP Owner Name: root

root
Reg. context: (not set)


Regexten on Qualify: No


Caller ID: Unknown

Unknown
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: 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


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: 127.0.0.1:5060


STUN server: 0.0.0.0:0

Global Signalling Settings:

Codecs: 0xc (ulaw|alaw)


Codec Order: ulaw:20,alaw:20


Relax DTMF: No


RFC2833 Compensation: No


Compact SIP headers: No


RTP Keepalive: 0 (Disabled) (Disabled)
RTP Timeout: 0 (Disabled) (Disabled)
RTP Hold Timeout: 0 (Disabled)


MWI NOTIFY mime type: application/simple-message-summary 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

No
Notify hold state: Yes


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: TCP,UDP


Outbound transport: UDP


Context: from-sip-external


Nat: RFC3581


DTMF: rfc2833


Qualify: 0


Use ClientCode: No


Progress inband: Never

Language:
Language:
MOH Interpret: default


MOH Suggest: Suggest:
Voice Mail Extension: *97*97

ip ro show

1x.6x.18.0/30 dev eth1 proto kernel scope link src 10.60.18.2ist item
4x.4x.15x.8x/29 dev eth2 proto kernel scope link src 46.46.153.90
192.168.0.0/22 dev eth0 proto kernel scope link src 192.168.0.38
169.254.0.0/16 dev eth2 scope link
default via 1x.60.18.1 dev eth1
default via 4x.4x.153.89 dev eth2eth2


asterisk -rx "sip show settings"
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: 5060
TCP Bindaddress: 0.0.0.0
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: Yes
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: Yes
Direct RTP setup: No
User Agent: FPBX-2.5.2(1.6.2.6)
SDP Session Name: Asterisk PBX 1.6.2.6
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
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: 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
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: 127.0.0.1:5060
STUN server: 0.0.0.0:0

Global Signalling Settings:

Codecs: 0xc (ulaw|alaw)
Codec Order: ulaw:20,alaw:20
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: Yes
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: TCP,UDP
Outbound transport: UDP
Context: from-sip-external
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

ip ro show

1x.6x.18.0/30 dev eth1 proto kernel scope link src 10.60.18.2ist item
item 4x.4x.15x.8x/29 dev eth2 proto kernel scope link src 46.46.153.90
46.46.153.90 192.168.0.0/22 dev eth0 proto kernel scope link src 192.168.0.38
192.168.0.38 169.254.0.0/16 dev eth2 scope link
default via 1x.60.18.1 dev eth1
default via 4x.4x.153.89 dev eth2

asterisk -rx "sip show settings"
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: 5060
TCP Bindaddress: 0.0.0.0
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: Yes
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: Yes
Direct RTP setup: No
User Agent: FPBX-2.5.2(1.6.2.6)
SDP Session Name: Asterisk PBX 1.6.2.6
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
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: 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
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: 127.0.0.1:5060
STUN server: 0.0.0.0:0

Global Signalling Settings:

Codecs: 0xc (ulaw|alaw)
Codec Order: ulaw:20,alaw:20
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: Yes
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: TCP,UDP
Outbound transport: UDP
Context: from-sip-external
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

ip ro show

show ---------- 1x.6x.18.0/30 dev eth1 proto kernel scope link src 10.60.18.2ist item 4x.4x.15x.8x/29 dev eth2 proto kernel scope link src 46.46.153.90 192.168.0.0/22 dev eth0 proto kernel scope link src 192.168.0.38 169.254.0.0/16 dev eth2 scope link
link default via 1x.60.18.1 dev eth1
eth1 default via 4x.4x.153.89 dev eth2

asterisk **asterisk -rx "sip show settings" settings"**
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: 5060
TCP Bindaddress: 0.0.0.0
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: Yes
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: Yes
Direct RTP setup: No
User Agent: FPBX-2.5.2(1.6.2.6)
SDP Session Name: Asterisk PBX 1.6.2.6
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
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:

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
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No

Network Settings:

Settings: --------------------------- SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externip: 0.0.0.0:0
Externrefresh: 10
Internal IP: 127.0.0.1:5060
STUN server: 0.0.0.0:0

Global Signalling Settings:

Settings: --------------------------- Codecs: 0xc (ulaw|alaw)
Codec Order: ulaw:20,alaw:20
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: Yes
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:

Settings: ----------------- Allowed transports: TCP,UDP
Outbound transport: UDP
Context: from-sip-external
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

ip ro show
----------
 1x.6x.18.0/30 dev eth1  proto kernel  scope link  src 10.60.18.2ist item
10.60.18.2
4x.4x.15x.8x/29 dev eth2  proto kernel  scope link  src 46.46.153.90
192.168.0.0/22 dev eth0  proto kernel  scope link  src 192.168.0.38
169.254.0.0/16 dev eth2  scope link
default via 1x.60.18.1 dev eth1
default via 4x.4x.153.89 dev eth2

**asterisk

asterisk -rx "sip show settings"** settings"


Global Settings:

 UDP SIP Port:           5060
5060 UDP Bindaddress: 0.0.0.0
0.0.0.0 TCP SIP Port: 5060
5060 TCP Bindaddress: 0.0.0.0
0.0.0.0 TLS SIP Port: Disabled
Disabled Videosupport: No
No Textsupport: No
No Ignore SDP sess. ver.: No
No AutoCreate Peer: No
No Match Auth Username: No
No Allow unknown access: Yes
Yes Allow subscriptions: Yes
Yes Allow overlap dialing: Yes
Yes Allow promsic. redir: No
No Enable call counters: No
No SIP domain support: No
No Realm. auth: No
No Our auth realm asterisk
asterisk Call to non-local dom.: Yes
Yes URI user is phone no: No
No Always auth rejects: Yes
Yes Direct RTP setup: No
No User Agent: FPBX-2.5.2(1.6.2.6)
FPBX-2.5.2(1.6.2.6) SDP Session Name: Asterisk PBX 1.6.2.6
1.6.2.6 SDP Owner Name: root
root Reg. context: (not set)
set) Regexten on Qualify: No
No Caller ID: Unknown
Unknown From: Domain:
Domain: Record SIP history: Off
Off Call Events: Off
Off Auth. Failure Events: Off
Off T.38 support: No
No T.38 EC mode: Unknown
Unknown T.38 MaxDtgrm: -1
-1 SIP realtime: Disabled
Disabled Qualify Freq : 60000 ms
ms Network QoS Settings: --------------------------- IP ToS SIP: CS3
CS3 IP ToS RTP audio: EF
EF IP ToS RTP video: AF41
AF41 IP ToS RTP text: CS0
CS0 802.1p CoS SIP: 4
4 802.1p CoS RTP audio: 5
5 802.1p CoS RTP video: 6
6 802.1p CoS RTP text: 5
5 Jitterbuffer enabled: No
No Jitterbuffer forced: No
No Jitterbuffer max size: -1
-1 Jitterbuffer resync: -1
-1 Jitterbuffer impl:
impl: Jitterbuffer log: No
No Network Settings: --------------------------- SIP address remapping: Disabled, no localnet list
list Externhost: <none>
<none> Externip: 0.0.0.0:0
0.0.0.0:0 Externrefresh: 10
10 Internal IP: 127.0.0.1:5060
127.0.0.1:5060 STUN server: 0.0.0.0:0
0.0.0.0:0 Global Signalling Settings: --------------------------- Codecs: 0xc (ulaw|alaw)
(ulaw|alaw) Codec Order: ulaw:20,alaw:20
ulaw:20,alaw:20 Relax DTMF: No
No RFC2833 Compensation: No
No Compact SIP headers: No
No RTP Keepalive: 0 (Disabled)
(Disabled) RTP Timeout: 0 (Disabled)
(Disabled) RTP Hold Timeout: 0 (Disabled)
(Disabled) MWI NOTIFY mime type: application/simple-message-summary
application/simple-message-summary DNS SRV lookup: Yes
Yes Pedantic SIP support: No
No Reg. min duration 60 secs
secs Reg. max duration: 3600 secs
secs Reg. default duration: 120 secs
secs Outbound reg. timeout: 20 secs
secs Outbound reg. attempts: 0
0 Notify ringing state: Yes
Yes Include CID: No
No Notify hold state: Yes
Yes SIP Transfer mode: open
open Max Call Bitrate: 384 kbps
kbps Auto-Framing: No
No Outb. proxy: <not set="">
set=""> Session Timers: Accept
Accept Session Refresher: uas
uas Session Expires: 1800 secs
secs Session Min-SE: 90 secs
secs Timer T1: 500
500 Timer T1 minimum: 100
100 Timer B: 32000
32000 No premature media: Yes
Yes Default Settings: ----------------- Allowed transports: TCP,UDP
TCP,UDP Outbound transport: UDP
UDP Context: from-sip-external
from-sip-external Nat: RFC3581
RFC3581 DTMF: rfc2833
rfc2833 Qualify: 0
0 Use ClientCode: No
No Progress inband: Never
Language:
Never Language: MOH Interpret: default
default MOH Suggest:
Suggest: Voice Mail Extension: *97
ip ro show
1x.6x.18.0/30 dev eth1  proto kernel  scope link  src 10.60.18.2
4x.4x.15x.8x/29 dev eth2  proto kernel  scope link  src 46.46.153.90
192.168.0.0/22 dev eth0  proto kernel  scope link  src 192.168.0.38
169.254.0.0/16 dev eth2  scope link
default via 1x.60.18.1 dev eth1
default via 4x.4x.153.89 dev eth2

asterisk -rx "sip show settings"

Global Settings:

  UDP SIP Port:           5060
  UDP Bindaddress:        0.0.0.0
  TCP SIP Port:           5060
  TCP Bindaddress:        0.0.0.0
  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:  Yes
  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:    Yes
  Direct RTP setup:       No
  User Agent:             FPBX-2.5.2(1.6.2.6)
  SDP Session Name:       Asterisk PBX 1.6.2.6
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              Unknown
  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:             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
  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:            127.0.0.1:5060
  STUN server:            0.0.0.0:0

Global Signalling Settings:
---------------------------
  Codecs:                 0xc (ulaw|alaw)
  Codec Order:            ulaw:20,alaw:20
  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:      Yes
  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:     TCP,UDP
  Outbound transport:     UDP
  Context:                from-sip-external
  Nat:                    RFC3581
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   *97
ip ro show
1x.6x.18.0/30 dev eth1  proto kernel  scope link  src 10.60.18.2
4x.4x.15x.8x/29 dev eth2  proto kernel  scope link  src 46.46.153.90
192.168.0.0/22 dev eth0  proto kernel  scope link  src 192.168.0.38
169.254.0.0/16 dev eth2  scope link
default via 1x.60.18.1 dev eth1
default via 4x.4x.153.89 dev eth2

asterisk -rx "sip show settings"

Global Settings:

  UDP SIP Port:           5060
  UDP Bindaddress:        0.0.0.0
  TCP SIP Port:           5060
  TCP Bindaddress:        0.0.0.0
  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:  Yes
  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:    Yes
  Direct RTP setup:       No
  User Agent:             FPBX-2.5.2(1.6.2.6)
  SDP Session Name:       Asterisk PBX 1.6.2.6
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              Unknown
  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 Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externip:               0.0.0.0:0
  Externrefresh:          10
  Internal IP:            127.0.0.1:5060
  STUN server:            0.0.0.0:0

Default Settings:
-----------------
  Allowed transports:     TCP,UDP
  Outbound transport:     UDP
  Context:                from-sip-external
  Nat:                    RFC3581
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   *97
ip ro show
1x.6x.18.0/30 dev eth1  proto kernel  scope link  src 10.60.18.2
4x.4x.15x.8x/29 dev eth2  proto kernel  scope link  src 46.46.153.90
192.168.0.0/22 dev eth0  proto kernel  scope link  src 192.168.0.38
169.254.0.0/16 dev eth2  scope link
default via 1x.60.18.1 dev eth1
default via 4x.4x.153.89 dev eth2

asterisk -rx "sip show settings"

Global Settings:

  UDP SIP Port:           5060
  UDP Bindaddress:        0.0.0.0
  TCP SIP Port:           5060
  TCP Bindaddress:        0.0.0.0
  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:  Yes
  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:    Yes
  Direct RTP setup:       No
  User Agent:             FPBX-2.5.2(1.6.2.6)
  SDP Session Name:       Asterisk PBX 1.6.2.6
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              Unknown
  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 Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externip:               0.0.0.0:0
  Externrefresh:          10
  Internal IP:            127.0.0.1:5060
  STUN server:            0.0.0.0:0

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.