1 | изначальная версия редактировать | |
sip show settings
2 | No.2 Revision редактировать |
sip show settings
3 | No.3 Revision редактировать |
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
4 | No.4 Revision редактировать |
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
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
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
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
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
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
5 | No.5 Revision редактировать |
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
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
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
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
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
6 | No.6 Revision редактировать |
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
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
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
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
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
7 | No.7 Revision редактировать |
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
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
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
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
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
8 | No.8 Revision редактировать |
ip ro show
1x.6x.18.0/30 dev eth1 proto kernel scope link src 10.60.18.2ist
itemitem 4x.4x.15x.8x/29 dev eth2 proto kernel scope link src
46.46.153.9046.46.153.90 192.168.0.0/22 dev eth0 proto kernel scope link src
192.168.0.38192.168.0.38 169.254.0.0/16 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 msNetwork 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: NoNetwork 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: 0.0.0.0:0Global 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: YesDefault 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
9 | No.9 Revision редактировать |
ip roshowshow ---------- 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 scopelinklink default via 1x.60.18.1 dev
eth1eth1 default via 4x.4x.153.89 dev eth2
asterisk**asterisk -rx "sip showsettings"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 msNetwork 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: NoNetwork
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: 0.0.0.0:0Global 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: YesDefault
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
10 | No.10 Revision редактировать |
ip ro show----------1x.6x.18.0/30 dev eth1 proto kernel scope link src10.60.18.2ist item10.60.18.2 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 showUDP SIP Port:
11 | No.11 Revision редактировать |
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 NetworkQoS 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 NetworkSettings: --------------------------- 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: 0.0.0.0:0Global 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: YesDefault 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
12 | No.12 Revision редактировать |
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 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: 0.0.0.0:0Default 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
13 | No.13 Revision редактировать |
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 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.