Подключение SPA3102 к Asterisk 11 по tls

[abrakadabr123]

Код: Выделить всё

#openssl s_client -showcerts -debug -connect 127.0.0.1:5061 -no_ssl2 -bugs
CONNECTED(00000003)
write to 0x17a8580 [0x17a8b10] (308 bytes => 308 (0x134))
0000 - 16 03 01 01 2f 01 00 01-2b 03 03 55 06 a9 10 15   ..../...+..U....
................................................... и.т.д
---
Certificate chain
 0 s:/CN=site.ru
   i:/CN=site.ru
-----BEGIN CERTIFICATE-----
-----END CERTIFICATE-----
---
Server certificate
subject=/CN=site.ru
issuer=/CN=site.ru
---
No client certificate CA names sent
---
SSL handshake has read 853 bytes and written 507 bytes
---
New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384
Server public key is 1024 bit
Secure Renegotiation IS supported
Compression: zlib compression
Expansion: zlib compression
SSL-Session:
    Protocol  : TLSv1.2
    Cipher    : AES256-GCM-SHA384
    Session-ID: 
    Session-ID-ctx: 
    Master-Key: 
    Key-Arg   : None
    PSK identity: None
    PSK identity hint: None
    SRP username: None
    TLS session ticket lifetime hint: 300 (seconds)
    TLS session ticket:
    .............................................
    Compression: 1 (zlib compression)
    Start Time: 1426499856
    Timeout   : 300 (sec)
    Verify return code: 18 (self signed certificate)
---

[ded]

Я правильно понимаю, что у вас так и есть везде там CN=site.ru?

[abrakadabr123]

Нет. Вместо site.ru указано реальное доменное имя. В панели управления днс зоной у регистратора существует запись типа А с указанием ip адреса сервера на котором установлен астериск.

[abrakadabr123]

Люди помогите. Неужели ни у кого нет идей.

[ded]

ssldump port 5061

[abrakadabr123]

Может я ошибаюсь, но дамп странный получился.

Код: Выделить всё

# ssldump port 5061
New TCP connection #1: ip spa(18442) <-> ip asterisk(5061)
1    63.0159 (63.0159)  C>S  TCP RST
New TCP connection #7: ip spa(18443) <-> ip asterisk(5061)

Астериск находится за NAT. Порты 0-65536 проброшены. У SPA белый адрес.

[ded]

вы различаете как-то посильные и непосильные задачи для себя? Вот, например, есть задачи, которые я не возмусь делать - нет знаний и опыта! Как в анекдоте:
- Рядовой Петров!
- Я!
- Поднять танк, живо!
- ....?
- То-то! А ты как думал? 50 тонн!

Тут показано как дебажить TLS соединения:
http://www.kamailio.org/wiki/tutorials/ ... -debugging
то, что там Kamailio - можно игнорировать.

Астериск находится за NAT. Порты 0-65536 проброшены. У SPA белый адрес.

Проблема в НАТе. Скорее всего - только UDP порты проброшены.
Это можно проверить. Например - подключиться по TLS с софтфона из локальной сети Астериска. Если дамп будет нормальный - проблема в НАТообразующем устройстве.

Но можно и проверить
netstat -nlp | grep 5061

А вообще бы я уже направил в платный суппорт (но это не я).

[abrakadabr123]

Главное желание. Если есть желание решить поставленную задачу, то придут и знания и опыт.
Все оказалось весьма прозаично. Очень помогла Ваша фраза

ca.crt и asterisk.crt - это должны быть РАЗНЫЕ файлы, и к каждому свой приватный ключ - ca.key и asterisk.key.

Создал клиетский сертификат с ключом и уже клиентский ключ использовал при создании сертификата и ключа для SPA. Железяка зарегистрировалась. Но возникла другая проблема при включенном SRTP не проходят взонки даже на внутренние номера.

Код: Выделить всё

[2015-03-17 17:38:02] WARNING[3563][C-00000022]: chan_sip.c:10440 process_sdp: Matched device setup to use SRTP, but request was not!

Если выключить SRTP звонки идут.

[ded]

nmap ip_addr_SPA3102 -p 5061 ?

[abrakadabr123]

Звонок на несуществующий номер 999999. SRTP включен.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

SIP read from TLS:ip_spa:18566 --->
INVITE sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-a8c21166
From: spaline1 <sip:605381@ip_asterisk>;tag=96dd0a528332c6d9o0
To: <sip:999999@ip_asterisk>
Remote-Party-ID: spaline1 <sip:605381@ip_asterisk:5061>;screen=yes;party=calling
Call-ID: 61007686-2a612835@ip_spa
CSeq: 101 INVITE
Max-Forwards: 70
Contact: spaline1 <sip:605381@ip_spa:18566;transport=tls>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 249
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 27268 27268 IN IP4 ip_spa
s=-
c=IN IP4 ip_spa
t=0 0
m=audio 10010 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 13 lines) ---
Sending to ip_spa:18566 (no NAT)
Sending to ip_spa:18566 (no NAT)
Using INVITE request as basis request - 61007686-2a612835@ip_spa
Found peer '605381' for '605381' from ip_spa:18566

<--- Reliably Transmitting (no NAT) to ip_spa:18566 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-a8c21166;received=ip_spa
From: spaline1 <sip:605381@ip_asterisk>;tag=96dd0a528332c6d9o0
To: <sip:999999@ip_asterisk>;tag=as72c5fe54
Call-ID: 61007686-2a612835@ip_spa
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b2d4ffd"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '61007686-2a612835@ip_spa' in 18496 ms (Method: INVITE)

<--- SIP read from TLS:ip_spa:18566 --->
ACK sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-a8c21166
From: spaline1 <sip:605381@ip_asterisk>;tag=96dd0a528332c6d9o0
To: <sip:999999@ip_asterisk>;tag=as72c5fe54
Call-ID: 61007686-2a612835@ip_spa
CSeq: 101 ACK
Max-Forwards: 70
Contact: spaline1 <sip:605381@ip_spa:18566;transport=tls>
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from TLS:ip_spa:18566 --->
INVITE sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-92fa7ed6
From: spaline1 <sip:605381@ip_asterisk>;tag=96dd0a528332c6d9o0
To: <sip:999999@ip_asterisk>
Remote-Party-ID: spaline1 <sip:605381@ip_asterisk:5061>;screen=yes;party=calling
Call-ID: 61007686-2a612835@ip_spa
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="605381",realm="asterisk",nonce="0b2d4ffd",uri="sip:999999@ip_asterisk:5061",algorithm=MD5,response="2f000d3523cf8dedcc249223f32c3c41"
Contact: spaline1 <sip:605381@ip_spa:18566;transport=tls>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 249
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 27268 27268 IN IP4 ip_spa
s=-
c=IN IP4 ip_spa
t=0 0
m=audio 10010 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (16 headers 13 lines) ---
Sending to ip_spa:18566 (no NAT)
Using INVITE request as basis request - 61007686-2a612835@ip_spa
Found peer '605381' for '605381' from ip_spa:18566
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[2015-03-18 10:35:10] NOTICE[2664][C-00000003]: chan_sip.c:10028 process_sdp: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 10010 RTP/AVP 8 100 101
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
[2015-03-18 10:35:10] WARNING[2664][C-00000003]: chan_sip.c:10440 process_sdp: Matched device setup to use SRTP, but request was not!

<--- Reliably Transmitting (no NAT) to ip_spa:18566 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-92fa7ed6;received=ip_spa
From: spaline1 <sip:605381@ip_asterisk>;tag=96dd0a528332c6d9o0
To: <sip:999999@ip_asterisk>;tag=as72c5fe54
Call-ID: 61007686-2a612835@ip_spa
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '61007686-2a612835@ip_spa' in 18496 ms (Method: INVITE)

<--- SIP read from TLS:ip_spa:18566 --->
ACK sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-92fa7ed6
From: spaline1 <sip:605381@ip_asterisk>;tag=96dd0a528332c6d9o0
To: <sip:999999@ip_asterisk>;tag=as72c5fe54
Call-ID: 61007686-2a612835@ip_spa
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="605381",realm="asterisk",nonce="0b2d4ffd",uri="sip:999999@ip_asterisk:5061",algorithm=MD5,response="2f000d3523cf8dedcc249223f32c3c41"
Contact: spaline1 <sip:605381@ip_spa:18566;transport=tls>
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

SRTP выключен. Проигрывается сообщение о неправильном наборе.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

<--- SIP read from TLS:ip_spa:18566 --->
INVITE sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-764a5cee
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>
Remote-Party-ID: spaline1 <sip:605381@ip_asterisk:5061>;screen=yes;party=calling
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 101 INVITE
Max-Forwards: 70
Contact: spaline1 <sip:605381@ip_spa:18566;transport=tls>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 251
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 123487 123487 IN IP4 ip_spa
s=-
c=IN IP4 ip_spa
t=0 0
m=audio 10004 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 13 lines) ---
Sending to ip_spa:18566 (no NAT)
Sending to ip_spa:18566 (no NAT)
Using INVITE request as basis request - 1f8413ae-14bb6508@ip_spa
Found peer '605381' for '605381' from ip_spa:18566

<--- Reliably Transmitting (no NAT) to ip_spa:18566 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-764a5cee;received=ip_spa
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>;tag=as14d860b9
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e430c27"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1f8413ae-14bb6508@ip_spa' in 19136 ms (Method: INVITE)

<--- SIP read from TLS:ip_spa:18566 --->
ACK sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-764a5cee
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>;tag=as14d860b9
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 101 ACK
Max-Forwards: 70
Contact: spaline1 <sip:605381@ip_spa:18566;transport=tls>
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from TLS:ip_spa:18566 --->
INVITE sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-adb6ad66
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>
Remote-Party-ID: spaline1 <sip:605381@ip_asterisk:5061>;screen=yes;party=calling
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="605381",realm="asterisk",nonce="2e430c27",uri="sip:999999@ip_asterisk:5061",algorithm=MD5,response="ec553dfe6691c0da7cf15c073ba314f3"
Contact: spaline1 <sip:605381@ip_spa:18566;transport=tls>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 251
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 123487 123487 IN IP4 ip_spa
s=-
c=IN IP4 ip_spa
t=0 0
m=audio 10004 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (16 headers 13 lines) ---
Sending to ip_spa:18566 (no NAT)
Using INVITE request as basis request - 1f8413ae-14bb6508@ip_spa
Found peer '605381' for '605381' from ip_spa:18566
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[2015-03-18 10:51:12] NOTICE[2664][C-00000005]: chan_sip.c:10028 process_sdp: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 10004 RTP/AVP 8 100 101
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|siren7|siren14), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port ip_spa:10004
Looking for 999999 in from-internal (domain ip_asterisk)
list_route: hop: <sip:605381@ip_spa:18566;transport=tls>

<--- Transmitting (no NAT) to ip_spa:18566 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-adb6ad66;received=ip_spa
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:999999@ip_asterisk:5061;transport=TLS>
Content-Length: 0


<------------>
    -- Executing [999999@from-internal:1] ResetCDR("SIP/605381-00000002", "") in new stack
    -- Executing [999999@from-internal:2] NoCDR("SIP/605381-00000002", "") in new stack
    -- Executing [999999@from-internal:3] Progress("SIP/605381-00000002", "") in new stack
Audio is at 10006
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to ip_spa:18566 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-adb6ad66;received=ip_spa
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>;tag=as540ce7b9
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:999999@ip_asterisk:5061;transport=TLS>
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 215736877 215736877 IN IP4 ip_asterisk
s=Asterisk PBX 11.16.0
c=IN IP4 ip_asterisk
t=0 0
m=audio 10006 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
    -- Executing [999999@from-internal:4] Wait("SIP/605381-00000002", "1") in new stack
       > 0x28db390 -- Probation passed - setting RTP source address to ip_spa:10004
    -- Executing [999999@from-internal:5] Progress("SIP/605381-00000002", "") in new stack
    -- Executing [999999@from-internal:6] Playback("SIP/605381-00000002", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <SIP/605381-00000002> Playing 'silence/1.alaw' (language 'ru')
    -- <SIP/605381-00000002> Playing 'cannot-complete-as-dialed.slin' (language 'ru')
    -- <SIP/605381-00000002> Playing 'check-number-dial-again.slin' (language 'ru')

<--- SIP read from TLS:ip_spa:18566 --->
CANCEL sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-adb6ad66
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 102 CANCEL
Max-Forwards: 70
Authorization: Digest username="605381",realm="asterisk",nonce="2e430c27",uri="sip:999999@ip_asterisk:5061",algorithm=MD5,response="eae39cbcd3c700ca47d843d50b8a1694"
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to ip_spa:18566 (no NAT)

<--- Reliably Transmitting (no NAT) to ip_spa:18566 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-adb6ad66;received=ip_spa
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>;tag=as540ce7b9
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to ip_spa:18566 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-adb6ad66;received=ip_spa
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>;tag=as540ce7b9
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 102 CANCEL
Server: FPBX-2.11.0(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (from-internal, 999999, 6) exited non-zero on 'SIP/605381-00000002'
    -- Executing [h@from-internal:1] Hangup("SIP/605381-00000002", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/605381-00000002'

<--- SIP read from TLS:ip_spa:18566 --->
ACK sip:999999@ip_asterisk:5061 SIP/2.0
Via: SIP/2.0/TLS ip_spa:18566;branch=z9hG4bK-adb6ad66
From: spaline1 <sip:605381@ip_asterisk>;tag=de57b3761d7307e0o0
To: <sip:999999@ip_asterisk>;tag=as540ce7b9
Call-ID: 1f8413ae-14bb6508@ip_spa
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="605381",realm="asterisk",nonce="2e430c27",uri="sip:999999@ip_asterisk:5061",algorithm=MD5,response="ec553dfe6691c0da7cf15c073ba314f3"
Contact: spaline1 <sip:605381@ip_spa:18566;transport=tls>
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

Код: Выделить всё

Nmap scan report for ip_spa
Host is up (0.26s latency).
PORT     STATE  SERVICE
5061/tcp closed sip-tls

Nmap scan report for ip_spa
Host is up (0.24s latency).
PORT      STATE  SERVICE
18566/tcp closed unknown