Не удается зарегить телефон на астериске.
pbx*CLI>
<--- SIP read from 89.200.111.222:16727 --->
REGISTER sip:85.15.100.100;user=phone SIP/2.0
Via: SIP/2.0/UDP 89.200.111.222:16727;branch=z9hG4bK9252470814204754865-312347
From: <sip:113@89.200.111.222:5060;user=phone>;tag=c0a80101-4c41c
To: <sip:113@85.15.100.100:5060;user=phone>
Call-ID: 990d8-c0a80101-5-3@89.200.111.222
CSeq: 1 REGISTER
Max-Forwards: 70
Expires: 0
Contact: *
User-Agent: THOMSON ST2022 hw2 fw4.68 00-14-7F-E1-FC-6D
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 89.200.111.222 : 16727 (no NAT)
<--- Transmitting (NAT) to 89.200.111.222:16727 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.200.111.222:16727;branch=z9hG4bK9252470814204754865-312347;received=89.200.111.222
From: <sip:113@89.200.111.222:5060;user=phone>;tag=c0a80101-4c41c
To: <sip:113@85.15.100.100:5060;user=phone>
Call-ID: 990d8-c0a80101-5-3@89.200.111.222
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:113@85.15.100.100>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 89.200.111.222:16727 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 89.200.111.222:16727;branch=z9hG4bK9252470814204754865-312347;received=89.200.111.222
From: <sip:113@89.200.111.222:5060;user=phone>;tag=c0a80101-4c41c
To: <sip:113@85.15.100.100:5060;user=phone>;tag=as2ca2fff2
Call-ID: 990d8-c0a80101-5-3@89.200.111.222
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="516a0076"
Content-Length: 0
вот как прописано в телефоне
вот пир в астериске
pbx*CLI> sip show peer 113
pbx*CLI>
* Name : 113
Secret : <Set>
MD5Secret : <Not set>
Context : internal
Subscr.Cont. : local-ext
Language : ru
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup : 1
Pickupgroup : 1
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 10
Dynamic : Yes
Callerid : "113" <113>
MaxCallBR : 384 kbps
Expire : -1
Insecure : invite
Nat : Always
ACL : No
T38 pt UDPTL : Yes
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: Yes
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : (Unspecified) Port 0
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 113
SIP Options : (none)
Codecs : 0x8 (alaw)
Codec Order : (alaw:20)
Auto-Framing: No
Status : UNKNOWN
Useragent :
Reg. Contact :
Телефон находить за NAT, сервер на выделенном IP.
