Регистрация gsm-шлюза AddPac AP-GS708

[phone2user]

Друзья, прошу помощи, все работало при следующих настройках около 6 месяцев.

Asterisk:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[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)

Роутер:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

!
! 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:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

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

Прошу указать, где я неправ? Планировалось регить шлюз на астере по ip, без логинов-паролей.

[ded]

У Вас в части
! SIP UA configuration.
!
sip-ua
user-register
sip-server 192.168.0.55
timeout treg 3600
remote-party-id
register e164

попытка регистрировать user-register на сервере, и регистрировать номера (обычно на портах FXS, а вы номера симок регистрируете) register e164, это не стыкуется с

Планировалось регить шлюз на астере по ip, без логинов-паролей.
....
т.к. регистрация отвалилась.

Если без регистраций, то эти строки убрать, а на Астериске указать
type=peer/friend
host=192.168.4.90
context=test-gsm
и т.д.
Вызовы делать Dial(SIP/gsm/${EXTEN})
включить sip set debug ip 192.168.4.90
и делать звонок

[awsswa]

вытаскиваете каждую симку, суёте в телефон и делаете звонок - с неё и на неё.

[phone2user]

Если без регистраций, то эти строки убрать, а на Астериске указать
type=peer/friend
host=192.168.4.90
context=test-gsm
и т.д.
Вызовы делать Dial(SIP/gsm/${EXTEN})
включить sip set debug ip 192.168.4.90
и делать звонок

На аддпаке сделал:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

sip-ua
sip-server 192.168.0.55 5060 126
timeout treg 3600
remote-party-id
hook-flash-info-ignore
!

На астере сделал:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[gsm-peers](!)
type=friend
host=192.168.4.90
context=test-gsm
dtmfmode=rfc2833
insecure=port,invite
qualify=yes
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)

в ответ на звонок на астер прилетает вот это:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- Transmitting (NAT) to 192.168.4.90:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.4.90:5060;branch=z9hG4bK4b5886fba4971;received=192.168.4.90;rport=5060
From: <sip:0T@192.168.0.55>;tag=4b5886fba4
To: sip:0T@192.168.0.55;tag=as5468a90a
Call-ID: 4be15858-6206-860e-80fb-0002a409c3a2@192.168.4.90
CSeq: 971 REGISTER
Server: Sever Kirov
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="077c6a27"
Content-Length: 0

я вижу свой косяк в поле From: <sip:0T@192.168.0.55>, это из-за destination-pattern 0T в настройках диалпиров, но если сделать в диалпире destination-pattern 89004951046, тогда опять исходящий на этот номер не будет работать.

[ded]

https://www.voipinfo.ru/forum/viewtopic.php?f=9&t=8765
https://www.voipinfo.ru/forum/viewforum.php?f=9