asterisk.ru

localhost*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
044XXXXXXX/044XXXXXXX 212.5xx.xxx.xx 5061 OK (1 ms)
Life 77.8x.xxx.xx 5060 OK (2 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
localhost*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
212.5xx.xxx.xx:5061 N 044XXXXXXX 105 Registered Mon, 10 Dec 2012 13:23:39
1 SIP registrations.
localhost*CLI>

[general]
transport=udp
tlsenable=yes ; Enable server for incoming TLS (secure) connections (default is no)
tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
rfc2833compensate=yes
;------------------------ TLS settings ------------------------------------------------------------
tlscertfile=/etc/asterisk/certificate/asterisk.pem
tlscafile=/etc/asterisk/certificate/ca.pem
tlscipher=DES-CBC3-SHA
tlsclientmethod=tlsv1
tlsdontverifyserver=no
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
register => tls://044XXXXXXX:verysecret@212.5X.XXX.XX:5061/044XXXXXXX

[044XXXXXXX] ; пир, SIP-канал в сторону провайдера. Создается для каждой номерной линии.
type=friend
username=044XXXXXXX
secret=verysecret
host=212.5X.XXX.X
port=5061
fromuser=044XXXXXXX
transport=TLS
encryption=yes
nat=no
context=incoming
qualify=yes
disallow=all
allow=ulaw
allow=alaw


[Life]
disallow=all
type=friend
context=outgoing
host=77.8X.XXX.XX
nat=no
encryption=no
canreinvite=no
qualify=yes
port=5060
call-limit=90
allow=alaw
insecure=invite,port

[incoming]
exten => _X.,1,Dial(Sip/Life/${EXTEN})
exten => _X.,n,Playback(vm-nobodyavail)
exten => _X.,n,Hangup()

[outgoing]
exten => _X.,1,Set(_SIP_SRTP_SDES=1)
exten => _X.,2,Set(_SIPSRTP=1)
exten => _X.,3,Set(_SIP_SRTP_CRYPTO=enable)
exten => _X.,4,NoOp(${CALLERID(name)})
exten => _X.,5,NoOp(${CALLERID(num)})
exten => _X.,6,Dial(Sip/044XXXXXXX/${EXTEN})
exten => _X.,n,Hangup()

== Using SIP RTP CoS mark 5
-- Executing [067502XXXX@outgoing:1] Set("SIP/Life-00000018", "_SIP_SRTP_SDES=1") in new stack
-- Executing [067502XXXX@outgoing:2] Set("SIP/Life-00000018", "_SIPSRTP=1") in new stack
-- Executing [067502XXXX@outgoing:3] Set("SIP/Life-00000018", "_SIP_SRTP_CRYPTO=enable") in new stack
-- Executing [067502XXXX@outgoing:4] NoOp("SIP/Life-00000018", "100292") in new stack
-- Executing [067502XXXX@outgoing:5] NoOp("SIP/Life-00000018", "100292") in new stack
-- Executing [067502XXXX@outgoing:6] Dial("SIP/Life-00000018", "Sip/044XXXXXXX/067502XXXX") in new stack
== Using SIP RTP CoS mark 5
Audio is at 11356
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.5X.XXX.XX:5061:
INVITE sip:067502XXXX@212.5X.XXX.XX:5061 SIP/2.0
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK411a9552
Max-Forwards: 70
From: "100292" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as58979715
To: <sip:067502XXXX@212.5X.XXX.XX:5061>
Contact: <sip:044XXXXXXX@195.2XX.XXX.XXX:5061;transport=TLS>
Call-ID: 2c2d17c97df83c347273a28a55724bf5@195.2XX.XXX.XXX:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Date: Mon, 10 Dec 2012 11:10:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 379

v=0
o=root 1567557296 1567557296 IN IP4 195.2XX.XXX.XXX
s=Asterisk PBX 1.8.11.0
c=IN IP4 195.2XX.XXX.XXX
t=0 0
m=audio 11356 RTP/SAVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hw0zS1/yDsTD9/PDmgg4MT4tYKvKcdgw7gFj+WNu

---
-- Called Sip/044XXXXXXX/067502XXXX

<--- SIP read from TLS:212.5X.XXX.XX:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK411a9552;received=195.2XX.XXX.XXX
From: "100292" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as58979715
To: <sip:067502XXXX@212.5X.XXX.XX:5061>;tag=as70e4e3ea
Call-ID: 2c2d17c97df83c347273a28a55724bf5@195.2XX.XXX.XXX:5061
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05a41e22"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:067502XXXX@212.5X.XXX.XX:5061> for address/port to send to
set_destination: set destination to 212.5X.XXX.XX:5061
Transmitting (no NAT) to 212.5X.XXX.XX:5061:
ACK sip:067502XXXX@212.5X.XXX.XX:5061 SIP/2.0
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK411a9552
Max-Forwards: 70
From: "100292" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as58979715
To: <sip:067502XXXX@212.5X.XXX.XX:5061>;tag=as70e4e3ea
Contact: <sip:044XXXXXXX@195.2XX.XXX.XXX:5061;transport=TLS>
Call-ID: 2c2d17c97df83c347273a28a55724bf5@195.2XX.XXX.XXX:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.11.0
Content-Length: 0


---
Audio is at 11356
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.5X.XXX.XX:5061:
INVITE sip:067502XXXX@212.5X.XXX.XX:5061 SIP/2.0
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK5df47af3
Max-Forwards: 70
From: "100292" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as58979715
To: <sip:067502XXXX@212.5X.XXX.XX:5061>
Contact: <sip:044XXXXXXX@195.2XX.XXX.XXX:5061;transport=TLS>
Call-ID: 2c2d17c97df83c347273a28a55724bf5@195.2XX.XXX.XXX:5061
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.11.0
Authorization: Digest username="044XXXXXXX", realm="asterisk", algorithm=MD5, uri="sip:067502XXXX@212.5X.XXX.XX:5061", nonce="05a41e22", response="b0e1870ade11026c136579dceca64f46"
Date: Mon, 10 Dec 2012 11:10:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 379

v=0
o=root 1567557296 1567557297 IN IP4 195.2XX.XXX.XXX
s=Asterisk PBX 1.8.11.0
c=IN IP4 195.2XX.XXX.XXX
t=0 0
m=audio 11356 RTP/SAVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hw0zS1/yDsTD9/PDmgg4MT4tYKvKcdgw7gFj+WNu

---

<--- SIP read from TLS:212.5X.XXX.XX:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK5df47af3;received=195.2XX.XXX.XXX
From: "100292" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as58979715
To: <sip:067502XXXX@212.5X.XXX.XX:5061>
Call-ID: 2c2d17c97df83c347273a28a55724bf5@195.2XX.XXX.XXX:5061
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:067502XXXX@212.5X.XXX.XX:5061;transport=TLS>
Content-Length: 0

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

<--- SIP read from TLS:212.5X.XXX.XX:5061 --->
SIP/2.0 500 Server internal failure
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK5df47af3;received=195.2XX.XXX.XXX
From: "100292" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as58979715
To: <sip:067502XXXX@212.5X.XXX.XX:5061>;tag=as70df7943
Call-ID: 2c2d17c97df83c347273a28a55724bf5@195.2XX.XXX.XXX:5061
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
-- Got SIP response 500 "Server internal failure" back from 212.5X.XXX.XX:5061
Transmitting (no NAT) to 212.5X.XXX.XX:5061:
ACK sip:067502XXXX@212.5X.XXX.XX:5061 SIP/2.0
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK5df47af3
Max-Forwards: 70
From: "100292" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as58979715
To: <sip:067502XXXX@212.5X.XXX.XX:5061>;tag=as70df7943
Contact: <sip:044XXXXXXX@195.2XX.XXX.XXX:5061;transport=TLS>
Call-ID: 2c2d17c97df83c347273a28a55724bf5@195.2XX.XXX.XXX:5061
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.11.0
Content-Length: 0


---
-- SIP/044XXXXXXX-00000019 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [067502XXXX@outgoing:7] Hangup("SIP/Life-00000018", "") in new stack
== Spawn extension (outgoing, 067502XXXX, 7) exited non-zero on 'SIP/Life-00000018'
Really destroying SIP dialog '2c2d17c97df83c347273a28a55724bf5@195.2XX.XXX.XXX:5061' Method: INVITE
Reliably Transmitting (no NAT) to 212.5X.XXX.XX:5061:
OPTIONS sip:212.5X.XXX.XX SIP/2.0
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK19aed31d
Max-Forwards: 70
From: "asterisk" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as6508932c
To: <sip:212.5X.XXX.XX>
Contact: <sip:044XXXXXXX@195.2XX.XXX.XXX:5061;transport=TLS>
Call-ID: 1e7b1b4b459c3d836d0d82696b1ec35f@195.2XX.XXX.XXX:5061
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Mon, 10 Dec 2012 11:10:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from TLS:212.5X.XXX.XX:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TLS 195.2XX.XXX.XXX:5061;branch=z9hG4bK19aed31d;received=195.2XX.XXX.XXX
From: "asterisk" <sip:044XXXXXXX@195.2XX.XXX.XXX>;tag=as6508932c
To: <sip:212.5X.XXX.XX>;tag=as0b993290
Call-ID: 1e7b1b4b459c3d836d0d82696b1ec35f@195.2XX.XXX.XXX:5061
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '1e7b1b4b459c3d836d0d82696b1ec35f@195.2XX.XXX.XXX:5061' Method: OPTIONS
localhost*CLI> sip set debug o
on off
localhost*CLI> sip set debug off
SIP Debugging Disabled
localhost*CLI>

Audio is at 5061
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 195.2XX.XXX.XXX:5061:
INVITE sip:s@195.2XX.XXX.XXX:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 212.5X.XXX.XX:5061;branch=z9hG4bK3d514e48
Max-Forwards: 70
From: "0442339000" <sip:0442339000@212.5X.XXX.XX>;tag=as208588f9
To: <sip:s@195.2XX.XXX.XXX:5061;transport=TLS>
Contact: <sip:0442339000@212.5X.XXX.XX:5061;transport=TLS>
Call-ID: 78524b305bfa50a34399c92278f9ea28@212.5X.XXX.XX:5061
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.7.1
Authorization: Digest username="044XXXXXXX", realm="asterisk", algorithm=MD5, uri="sip:s@195.2XX.XXX.XXX:5061;transport=TLS", nonce="7ab840f3", response="d7a92589f2857ea4588078cb2be77b16"
Date: Fri, 07 Dec 2012 08:54:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 374

v=0
o=root 1218337419 1218337420 IN IP4 212.5X.XXX.XX
s=Asterisk PBX 1.8.7.1
c=IN IP4 212.5X.XXX.XX
t=0 0
m=audio 11296 RTP/SAVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:iHGLqyzjkrSQHygwB5iS/9sH6De6Jwsovr7n7ZBZ

---

<--- SIP read from TLS:195.2XX.XXX.XXX:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TLS 212.5X.XXX.XX:5061;branch=z9hG4bK3d514e48;received=212.5X.XXX.XX
From: "0442339000" <sip:0442339000@212.5X.XXX.XX>;tag=as208588f9
To: <sip:s@195.2XX.XXX.XXX:5061;transport=TLS>;tag=as196f8bb6
Call-ID: 78524b305bfa50a34399c92278f9ea28@212.5X.XXX.XX:5061
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 195.2XX.XXX.XXX:5061:
ACK sip:s@195.2XX.XXX.XXX:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 212.5X.XXX.XX:5061;branch=z9hG4bK3d514e48
Max-Forwards: 70
From: "0442339000" <sip:0442339000@212.5X.XXX.XX>;tag=as208588f9
To: <sip:s@195.2XX.XXX.XXX:5061;transport=TLS>;tag=as196f8bb6
Contact: <sip:0442339000@212.5X.XXX.XX:5061;transport=TLS>
Call-ID: 78524b305bfa50a34399c92278f9ea28@212.5X.XXX.XX:5061
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.7.1
Content-Length: 0

* Name : 04423ХХХХХ
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : incoming
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromUser : 04423ХХХХХ
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : No
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 212.5х.ххх.хх
Addr->IP : 212.5х.ххх.хх:5061
Defaddr->IP : (null)
Prim.Transp. : TLS
Allowed.Trsp : TLS
Def. Username: 04423ХХХХХ
SIP Options : replaces replace timer
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing : No
Status : OK (1 ms)
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : Yes