Asterisk:
[Показать] Спойлер:
[gsm]
type=friend
call-limit=8
host=192.168.4.90
port=5060
nat=no
qualify=no
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=alaw
context=test-gsm
[gsm-peers](!)
type=friend
host=dynamic
;host=192.168.4.90
context=test-gsm
dtmfmode=inband
insecure=port,invite
qualify=no
deny=0.0.0.0/0.0.0.0
permit=192.168.4.90/255.255.0.0
nat=no
trustrpid=yes
[89004951116](gsm-peers)
[89004951081](gsm-peers)
[89004951058](gsm-peers)
[89004951046](gsm-peers)
[89004951045](gsm-peers)
[89004951198](gsm-peers)
[89004951161](gsm-peers)
[89004951142](gsm-peers)
type=friend
call-limit=8
host=192.168.4.90
port=5060
nat=no
qualify=no
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=alaw
context=test-gsm
[gsm-peers](!)
type=friend
host=dynamic
;host=192.168.4.90
context=test-gsm
dtmfmode=inband
insecure=port,invite
qualify=no
deny=0.0.0.0/0.0.0.0
permit=192.168.4.90/255.255.0.0
nat=no
trustrpid=yes
[89004951116](gsm-peers)
[89004951081](gsm-peers)
[89004951058](gsm-peers)
[89004951046](gsm-peers)
[89004951045](gsm-peers)
[89004951198](gsm-peers)
[89004951161](gsm-peers)
[89004951142](gsm-peers)
Роутер:
[Показать] Спойлер:
!
! APOS(tm) configuration saved from vty
! 2016/04/14 10:45:26
!
version 8.51.010
!
hostname MSK98A-GSM
clock timezone MSK 3
!
username root password router administrator
username guest password guest user
!
!
script ntpdate default
server ip 192.168.0.240
server ip 192.168.0.241
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 192.168.4.90 255.255.0.0
speed auto
no qos-control
!
interface FastEthernet0/1
no ip address
shutdown
speed auto
no qos-control
!
!
snmp community public ro 0.0.0.0
!
!
snmp server
http server
!
utilization cpu
utilization FastEthernet0/0
utilization FastEthernet0/1
!
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
protocol sip
dtmf-relay rfc-2833
fax protocol bypass
fax rate 9600
h323 call start fast
h323 call tunnel enable
no call-barring unconfigured-ip-address
no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
input gain 1
output gain 1
connection plar 89004951116
description 89004951116
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 0/1
input gain 1
output gain 1
connection plar 89004951081
description 89004951081
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 0/2
input gain 1
output gain 1
connection plar 89004951058
description 89004951058
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 0/3
input gain 1
output gain 1
connection plar 89004951046
description 89004951046
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 1/0
input gain 1
output gain 1
connection plar 89004951045
description 89004951045
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 1/1
input gain 1
output gain 1
connection plar 89004951198
description 89004951198
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 1/2
input gain 1
output gain 1
connection plar 89004951161
description 89004951161
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 1/3
input gain 1
output gain 1
connection plar 89004951142
description 89004951142
dial-tone-generate
caller-id enable
caller-id name disable
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
destination-pattern 0T
port 0/0
translate-outgoing called-number 9
!
dial-peer voice 2 pots
destination-pattern 0T
port 0/1
translate-outgoing called-number 9
!
dial-peer voice 3 pots
destination-pattern 0T
port 0/2
translate-outgoing called-number 9
!
dial-peer voice 4 pots
destination-pattern 0T
port 0/3
translate-outgoing called-number 9
!
dial-peer voice 5 pots
destination-pattern 0T
port 1/0
translate-outgoing called-number 9
!
dial-peer voice 6 pots
destination-pattern 0T
port 1/1
translate-outgoing called-number 9
!
dial-peer voice 7 pots
destination-pattern 0T
port 1/2
translate-outgoing called-number 9
!
dial-peer voice 8 pots
destination-pattern 0T
port 1/3
translate-outgoing called-number 9
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 0
no vad
dtmf-relay rtp-2833
!
!
!
!
dial-peer hunt 7
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.4.90
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
codec preference 4 g7231r63
!
!
!
! Translation Rule configuration.
!
translation-rule 9
rule 0 0T T
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-server 192.168.0.55
timeout treg 3600
remote-party-id
register e164
hook-flash-info-ignore
!
!
! Tones
voice class dial-tone 425 0 10000 0 0 0 -12
!
voice class ring-back-tone 425 0 600 2400 0 0 -12
!
voice class line-busy-tone 425 0 300 300 0 0 -12
!
voice class reorder-tone 430 0 300 300 0 0 -12
!
voice class line-lock-tone 425 0 350 350 0 0 -12
!
!
!
! SMS delivery configuration
!
sms-delivery
!
!
!
!
voip-interface ip FastEthernet0/0
!
line console
!
line vty
!
character-set encoding usa ascii
! mount mem 1024 /tmp
!
mobile dev-restart-by-unreg 180
mobile dev-restart-by-unknown-error
mobile cell-monitor 30
!
mobile 0/0
gsm sms-language utf8
!
mobile 0/1
gsm sms-language utf8
!
mobile 0/2
gsm sms-language utf8
!
mobile 0/3
gsm sms-language utf8
!
mobile 1/0
gsm sms-language utf8
!
mobile 1/1
gsm sms-language utf8
!
mobile 1/2
gsm sms-language utf8
!
mobile 1/3
gsm sms-language utf8
!
! APOS(tm) configuration saved from vty
! 2016/04/14 10:45:26
!
version 8.51.010
!
hostname MSK98A-GSM
clock timezone MSK 3
!
username root password router administrator
username guest password guest user
!
!
script ntpdate default
server ip 192.168.0.240
server ip 192.168.0.241
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 192.168.4.90 255.255.0.0
speed auto
no qos-control
!
interface FastEthernet0/1
no ip address
shutdown
speed auto
no qos-control
!
!
snmp community public ro 0.0.0.0
!
!
snmp server
http server
!
utilization cpu
utilization FastEthernet0/0
utilization FastEthernet0/1
!
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
protocol sip
dtmf-relay rfc-2833
fax protocol bypass
fax rate 9600
h323 call start fast
h323 call tunnel enable
no call-barring unconfigured-ip-address
no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
input gain 1
output gain 1
connection plar 89004951116
description 89004951116
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 0/1
input gain 1
output gain 1
connection plar 89004951081
description 89004951081
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 0/2
input gain 1
output gain 1
connection plar 89004951058
description 89004951058
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 0/3
input gain 1
output gain 1
connection plar 89004951046
description 89004951046
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 1/0
input gain 1
output gain 1
connection plar 89004951045
description 89004951045
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 1/1
input gain 1
output gain 1
connection plar 89004951198
description 89004951198
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 1/2
input gain 1
output gain 1
connection plar 89004951161
description 89004951161
dial-tone-generate
caller-id enable
caller-id name disable
!
!
! GSM
voice-port 1/3
input gain 1
output gain 1
connection plar 89004951142
description 89004951142
dial-tone-generate
caller-id enable
caller-id name disable
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
destination-pattern 0T
port 0/0
translate-outgoing called-number 9
!
dial-peer voice 2 pots
destination-pattern 0T
port 0/1
translate-outgoing called-number 9
!
dial-peer voice 3 pots
destination-pattern 0T
port 0/2
translate-outgoing called-number 9
!
dial-peer voice 4 pots
destination-pattern 0T
port 0/3
translate-outgoing called-number 9
!
dial-peer voice 5 pots
destination-pattern 0T
port 1/0
translate-outgoing called-number 9
!
dial-peer voice 6 pots
destination-pattern 0T
port 1/1
translate-outgoing called-number 9
!
dial-peer voice 7 pots
destination-pattern 0T
port 1/2
translate-outgoing called-number 9
!
dial-peer voice 8 pots
destination-pattern 0T
port 1/3
translate-outgoing called-number 9
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 0
no vad
dtmf-relay rtp-2833
!
!
!
!
dial-peer hunt 7
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.4.90
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
codec preference 4 g7231r63
!
!
!
! Translation Rule configuration.
!
translation-rule 9
rule 0 0T T
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-server 192.168.0.55
timeout treg 3600
remote-party-id
register e164
hook-flash-info-ignore
!
!
! Tones
voice class dial-tone 425 0 10000 0 0 0 -12
!
voice class ring-back-tone 425 0 600 2400 0 0 -12
!
voice class line-busy-tone 425 0 300 300 0 0 -12
!
voice class reorder-tone 430 0 300 300 0 0 -12
!
voice class line-lock-tone 425 0 350 350 0 0 -12
!
!
!
! SMS delivery configuration
!
sms-delivery
!
!
!
!
voip-interface ip FastEthernet0/0
!
line console
!
line vty
!
character-set encoding usa ascii
! mount mem 1024 /tmp
!
mobile dev-restart-by-unreg 180
mobile dev-restart-by-unknown-error
mobile cell-monitor 30
!
mobile 0/0
gsm sms-language utf8
!
mobile 0/1
gsm sms-language utf8
!
mobile 0/2
gsm sms-language utf8
!
mobile 0/3
gsm sms-language utf8
!
mobile 1/0
gsm sms-language utf8
!
mobile 1/1
gsm sms-language utf8
!
mobile 1/2
gsm sms-language utf8
!
mobile 1/3
gsm sms-language utf8
!
Поступили жалобы на обрывы связи, погрешив на большой аптайм я пустил в ребут addpac.
Исходящие сейчас работают, входящие перестали, т.к. регистрация отвалилась.
Инфа с аддпака:
[Показать] Спойлер:
show sip
Proxyserver Registration Information
proxyserver registration option = gateway
Proxyserver list :
---------------------------------------------------------------------------
Server address Port Priority Domain Status(LastFailReason)
---------------------------------------------------------------------------
192.168.0.55 5060 126 any Failed(Auth:noLocalAuthInfo)
Proxy Server registration status :
-----------------------------------------------------------------------------------
E.164 UserName Password Port Status
-----------------------------------------------------------------------------------
SIP UA Timer counters
retry counter = 10
SIP UA Timer values
tretry (sip retry timer) = 500 msec.
tinterval (sip retry max interval timer) = 4 sec.
treg (sip register timer) = 3600 sec.
tregtry (sip register retry timer) = 20 sec.
texpires (sip invite expire timer) = 180 sec.
tsipping (sip ping timer) = 45 sec.
tsrv (sip srv retry timer) = 60 sec.
SIP UA Session Timer value
Min-SE = 1800 sec.
Session-Expires = 1800 sec.
SIP DNS SRV Query : Disable
SIP Called-Party-Number : from URL
SIP Call Transfer Mode : Basic
SIP Media Channel Start Mode : Default
SIP Reliable Provisional Response Option : Supported with value <100rel>
SIP Response Option : default
SIP Local Domain : NULL
SIP Special Char : NULL
SIP Routing Method of Incoming Call : Default
SIP Remote-Party-ID : Enabled
SIP Local Host Name : No
SIP Conference Server Info
Name (ID) = NULL
Related Voip Tag = -1
SIP NAT Info
PING = Disabled
Required = NULL
SIP Session Refresh Method = INVITE
SIP Keep Authentication information on registration = Yes
SIP Message Parameter Translation TRUE
SIP Force-Forwarding Info
SIP Hook-Flash Event(INFO) Ignore = TRUE
SIP Time Sync With REGISTER Msg = FALSE
Proxyserver Registration Information
proxyserver registration option = gateway
Proxyserver list :
---------------------------------------------------------------------------
Server address Port Priority Domain Status(LastFailReason)
---------------------------------------------------------------------------
192.168.0.55 5060 126 any Failed(Auth:noLocalAuthInfo)
Proxy Server registration status :
-----------------------------------------------------------------------------------
E.164 UserName Password Port Status
-----------------------------------------------------------------------------------
SIP UA Timer counters
retry counter = 10
SIP UA Timer values
tretry (sip retry timer) = 500 msec.
tinterval (sip retry max interval timer) = 4 sec.
treg (sip register timer) = 3600 sec.
tregtry (sip register retry timer) = 20 sec.
texpires (sip invite expire timer) = 180 sec.
tsipping (sip ping timer) = 45 sec.
tsrv (sip srv retry timer) = 60 sec.
SIP UA Session Timer value
Min-SE = 1800 sec.
Session-Expires = 1800 sec.
SIP DNS SRV Query : Disable
SIP Called-Party-Number : from URL
SIP Call Transfer Mode : Basic
SIP Media Channel Start Mode : Default
SIP Reliable Provisional Response Option : Supported with value <100rel>
SIP Response Option : default
SIP Local Domain : NULL
SIP Special Char : NULL
SIP Routing Method of Incoming Call : Default
SIP Remote-Party-ID : Enabled
SIP Local Host Name : No
SIP Conference Server Info
Name (ID) = NULL
Related Voip Tag = -1
SIP NAT Info
PING = Disabled
Required = NULL
SIP Session Refresh Method = INVITE
SIP Keep Authentication information on registration = Yes
SIP Message Parameter Translation TRUE
SIP Force-Forwarding Info
SIP Hook-Flash Event(INFO) Ignore = TRUE
SIP Time Sync With REGISTER Msg = FALSE
Прошу указать, где я неправ? Планировалось регить шлюз на астере по ip, без логинов-паролей.
