История изменений - Asterisk: Вопросы и Ответы

1

изначальная версия

редактировать

ответил Jun 13 '12

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 ~~eth2~~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 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 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

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