Askozia. Why can't I connect to an ISP?

Question

Asterisk

Hello everyone
All day long I've been struggling with setting up access to the provider.
IP PBX Askozia is used.
The connection with the provider is not established.
I post the conclusions of sip set debug.
Here is what I send to the provider:

spoiler

Really destroying SIP dialog '[email protected]' Met                                                                                                                     hod: REGISTER
Retransmitting #1 (NAT) to 109.94.93.119:5060:
REGISTER sip:trx-cps.voip.an-net.ru SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK5afee836;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as51db76e1
To: <sip:[email protected]>
Call-ID: 6096b06c400d[email protected]
CSeq: 112 REGISTER
User-Agent: AskoziaPBX
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0
---
Retransmitting #2 (NAT) to 109.94.93.119:5060:
REGISTER sip:trx-cps.voip.an-net.ru SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK5afee836;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as51db76e1
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 112 REGISTER
User-Agent: AskoziaPBX
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Retransmitting #3 (NAT) to 109.94.93.119:5060:
REGISTER sip:trx-cps.voip.an-net.ru SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK5afee836;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as51db76e1
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 112 REGISTER
User-Agent: AskoziaPBX
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Retransmitting #4 (NAT) to 109.94.93.119:5060:
REGISTER sip:trx-cps.voip.an-net.ru SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK5afee836;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as51db76e1
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 112 REGISTER
User-Agent: AskoziaPBX
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---
Reliably Transmitting (NAT) to 109.94.93.119:5060:
OPTIONS sip:trx-cps.voip.an-net.ru SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK064de1f0;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as6469d5b6
To: <sip:trx-cps.voip.an-net.ru>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: AskoziaPBX
Date: Fri, 29 May 2015 09:47:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                                                     H
Supported: replaces, timer
Content-Length: 0

What does the provider tell me?

spoiler

<--- Transmitting (NAT) to 79.111.156.4:5060 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK0982817f;received=79.111.156.4;rport=5060

From: <sip:[email protected]>;tag=as163c3ca5

To: <sip:[email protected]>;tag=as52eb545b

Call-ID: [email protected]

CSeq: 102 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63e0793f"

Content-Length: 0

According to the provider, they do not receive register data, although I explicitly specified them in the settings: register => 27005:[email protected]/27005.
Who knows what else could be?
Already tried a bunch of different settings. I set it up according to this article .
Moreover, a regular IP phone connects to this provider.
sip.conf settings for provider

spoiler

[general]
context=public-direct-dial
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
useragent=AskoziaPBX
relaxdtmf=yes
alwaysauthreject=yes
videosupport=yes
notifybusy=yes
counteronpeer=yes
notifyhold=no
pedantic=yes
callcounter=yes
snomautoprovisioningurl=http://192.168.1.25:56080/prov.php?mac={mac}
yealinkautoprovisioningurl=http://192.168.1.25:56080/
gigasetautoprovisioningurl=http://192.168.1.25:56080/gigaset

externip=xxxx
localnet=192.168.1.0/24


; Аннет
register => login:[email protected]:5060/login

; Аннет
[SIP-PROVIDER-85369491755682a218725a]
type=friend
defaultuser=login
secret=pass
fromuser=login
host=trx-cps.voip.an-net.ru
port=5060
context=SIP-PROVIDER-85369491755682a218725a-incoming
fromdomain=trx-cps.voip.an-net.ru
language=ru-ru
nat=yes
qualifyfreq=60
qualify=2000
directmedia=no
insecure=port,invite
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw
allow=alaw
allow=gsm

Answer 1

If you want help, describe the scheme for connecting your phone to the network in great detail (switches, routers and of course NAT with ip addresses) and attach tcpdump for wireshark