Возникла необходимость перейти на астер 16. Перенес конфиги со старой машины. И столкнулся с проблемой - проходит регистрация транков только от одного провайдера.
- Код: выделить все
Really destroying SIP dialog '1a17dd9b6cdcf85819e4a9c0288d6eab@93.190.242.54:5060' Method: OPTIONS
Retransmitting #1 (no NAT) to 217.хх.хх.17:5060:
REGISTER sip:217.хх.хх.17 SIP/2.0
Via: SIP/2.0/UDP 93.хх.хх.55:5060;branch=z9hG4bK3accbb45
Max-Forwards: 70
From: <sip:707хх.хх118@217.хх.хх.17>;tag=as481563ee
To: <sip:707хх.хх118@217.хх.хх.17>
Call-ID: 4adc42c822f641a94cf405b543267ed0@10.10.30.41
CSeq: 231 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX certified/16.8-cert2
Expires: 3600
Contact: <sip:707хх.хх118@93.хх.хх.51:5060>
Content-Length: 0
---
Retransmitting #2 (no NAT) to 217.76.71.17:5060:
REGISTER sip:217.76.71.17 SIP/2.0
Via: SIP/2.0/UDP 93.хх.хх.51:5060;branch=z9hG4bK3accbb45
Max-Forwards: 70
From: <sip:707хх.хх18@217.хх.хх.17>;tag=as481563ee
To: <sip:707хх.хх19@217.хх.хх.17>
Call-ID: 4adc42c822f641a94cf405b543267ed0@10.10.40.41
CSeq: 231 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX certified/16.8-cert2
Expires: 3600
Contact: <sip:707хх.хх18@93.хх.хх.55:5060>
Content-Length: 0
---
Reliably Transmitting (NAT) to 217.76.71.17:5060:
OPTIONS sip:217.76.71.17 SIP/2.0
Via: SIP/2.0/UDP 93.хх.хх.55:5060;branch=z9hG4bK5735b3b7;rport
Max-Forwards: 70
From: "asterisk" <sip:701хх.хх444@93.хх.хх.55>;tag=as4f0b3f70
To: <sip:217.76.71.17>
Contact: <sip:701хх.хх4@93.хх.хх.55:5060>
Call-ID: 620152326f431c6716e31c92667592cb@93.хх.хх.55:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX certified/16.8-cert2
Date: Tue, 27 Oct 2020 11:31:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #1 (NAT) to 217.хх.хх17:5060:
OPTIONS sip:217.76.71.17 SIP/2.0
Via: SIP/2.0/UDP 93.хх.хх.55:5060;branch=z9hG4bK5735b3b7;rport
Max-Forwards: 70
From: "asterisk" <sip:7010654444@93.хх.хх.55>;tag=as4f0b3f70
To: <sip:217.хх.хх.17>
Contact: <sip:7010хх.хх4@93.хх.хх.55:5060>
Call-ID: 620152326f431c6716e31c92667592cb@93.хх.хх.55:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX certified/16.8-cert2
Date: Tue, 27 Oct 2020 11:31:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
sip.conf
[7073468118]
disallow=all
allow=ulaw
username=707хх.хх118
type=friend
callcounter=yes
callbackextension=707хх.хх118
;careinvite=no
secret=UYjJ$e1v
nat=no
fromdomain=217.хх.хх.17
outboundproxy=217.хх.хх.17
insecure=port,invite
host=217.хх.хх.17
port=5060
qualify=yes
dtmfmode=RFC2833
fromuser=7073468118
context=nps
Был бы признателен за подсказку )
