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спросил 2014-05-08 10:04:26 +0400

tw1911 Gravatar tw1911

sip trunk между asterisk

Есть пример из книжки Asterisk: the definitive guide 2013 года. Но не получается. Два однотипных Asterisk 11.8.1 Поставленных с нуля.

ServerA: 192.168.0.177

sip.conf

[general]
register => ServerA:apples@192.168.0.175/serverB
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.177
tcpenable=no
[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia         ; assume device is behind NAT
dtmfmode=auto 
disallow=all
allow=g722
allow=ulaw
allow=alaw

[0000FFFF0001](office-phone)
secret=aaaaVVV2222

[0000FFFF0002](office-phone)
secret=bbbbVVV2222

[ServerB]
type=friend
host=192.168.0.175
port=5060
defaultuser=ServerA
secret=apples
context=incoming
disallow=all
allow = ulaw

extensions.conf

[LocalSets]
exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _6XXX,1,Dial(SIP/ServerB/${EXTEN:1})

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

ServerB: 192.168.0.175

sip.conf

[general]
register => ServerB:apples@192.168.0.177/serverA
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.175
tcpenable=no

[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw 
allow=alaw

[0000FFFF0001](office-phone)
secret=aaaaVVV2222

[0000FFFF0002](office-phone)
secret=bbbbVVV2222

[ServerA]
type=friend
host=192.168.0.177
port=5060
defaultuser=ServerB
secret=apples
context=incoming
disallow=all
allow = ulaw

extensions.conf

[LocalSets]

exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _5XXX,1,Dial(SIP/${EXTEN:1}@ServerA)

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

При наборе на софтфоне с сервера A 6101 получаем: 

    -- Executing [6101@LocalSets:1] Dial("SIP/0000FFFF0001-0000001d", "SIP/ServerB/101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ServerB/101
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/0000FFFF0001-0000001d' status is 'CHANUNAVAIL'

sip show registry

asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
192.168.0.175:5060                      N      ServerA            120 No Authentication
1 SIP registrations.

В общем нужна помощь. Ip tables отключен, звонки на локальные клиенты проходят, никаких натов нет, айпи статические, хотя и висят физически на одной машине в виртуалках, сквозной пинг проходит.

sip trunk между asterisk

Есть пример из книжки Asterisk: the definitive guide 2013 года. Но не получается. Два однотипных Asterisk 11.8.1 Поставленных с нуля.

ServerA: 192.168.0.177

sip.conf

[general]
register => ServerA:apples@192.168.0.175/serverB
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.177
tcpenable=no
[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia         ; assume device is behind NAT
dtmfmode=auto 
disallow=all
allow=g722
allow=ulaw
allow=alaw

[0000FFFF0001](office-phone)
secret=aaaaVVV2222

[0000FFFF0002](office-phone)
secret=bbbbVVV2222

[ServerB]
type=friend
host=192.168.0.175
port=5060
defaultuser=ServerA
secret=apples
context=incoming
disallow=all
allow = ulaw

extensions.conf

[LocalSets]
exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _6XXX,1,Dial(SIP/ServerB/${EXTEN:1})

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

ServerB: 192.168.0.175

sip.conf

[general]
register => ServerB:apples@192.168.0.177/serverA
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.175
tcpenable=no

[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw 
allow=alaw

[0000FFFF0001](office-phone)
secret=aaaaVVV2222

[0000FFFF0002](office-phone)
secret=bbbbVVV2222

[ServerA]
type=friend
host=192.168.0.177
port=5060
defaultuser=ServerB
secret=apples
context=incoming
disallow=all
allow = ulaw

extensions.conf

[LocalSets]

exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _5XXX,1,Dial(SIP/${EXTEN:1}@ServerA)

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

При наборе на софтфоне с сервера A 6101 получаем: 

    -- Executing [6101@LocalSets:1] Dial("SIP/0000FFFF0001-0000001d", "SIP/ServerB/101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ServerB/101
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/0000FFFF0001-0000001d' status is 'CHANUNAVAIL'

sip show registry

asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
192.168.0.175:5060                      N      ServerA            120 No Authentication
1 SIP registrations.

В общем нужна помощь. Ip tables отключен, звонки на локальные клиенты проходят, никаких натов нет, айпи статические, хотя и висят физически на одной машине в виртуалках, сквозной пинг проходит. Ответ awsswa не помог.

sip trunk между asterisk

Есть пример из книжки Asterisk: the definitive guide 2013 года. Но не получается. Два однотипных Asterisk 11.8.1 Поставленных с нуля.

ServerA: 192.168.0.177

sip.conf

[general]
register => ServerA:apples@192.168.0.175/serverB
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.177
tcpenable=no
[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia         ; assume device is behind NAT
dtmfmode=auto 
disallow=all
allow=g722
allow=ulaw
allow=alaw

[0000FFFF0001](office-phone)
secret=aaaaVVV2222

[0000FFFF0002](office-phone)
secret=bbbbVVV2222

[ServerB]
type=friend
host=192.168.0.175
port=5060
defaultuser=ServerA
secret=apples
context=incoming
disallow=all
allow = ulaw

extensions.conf

[LocalSets]
exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _6XXX,1,Dial(SIP/ServerB/${EXTEN:1})

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

ServerB: 192.168.0.175

sip.conf

[general]
register => ServerB:apples@192.168.0.177/serverA
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.175
tcpenable=no

[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw 
allow=alaw

[0000FFFF0001](office-phone)
secret=aaaaVVV2222

[0000FFFF0002](office-phone)
secret=bbbbVVV2222

[ServerA]
type=friend
host=192.168.0.177
port=5060
defaultuser=ServerB
secret=apples
context=incoming
disallow=all
allow = ulaw

extensions.conf

[LocalSets]

exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _5XXX,1,Dial(SIP/${EXTEN:1}@ServerA)

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

При наборе на софтфоне с сервера A 6101 получаем: 

    -- Executing [6101@LocalSets:1] Dial("SIP/0000FFFF0001-0000001d", "SIP/ServerB/101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ServerB/101
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/0000FFFF0001-0000001d' status is 'CHANUNAVAIL'

sip show registry

asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
192.168.0.175:5060                      N      ServerA            120 No Authentication
1 SIP registrations.

В общем нужна помощь. Ip tables отключен, звонки на локальные клиенты проходят, никаких натов нет, айпи статические, хотя и висят физически на одной машине в виртуалках, сквозной пинг проходит. Ответ awsswa не помог. zzuz Ладно, хорошо, почему текущий конфиг не работает вообще?

sip trunk между asterisk

Есть пример из книжки Asterisk: the definitive guide 2013 года. Но не получается. Два однотипных Asterisk 11.8.1 Поставленных с нуля.

ServerA: 192.168.0.177

sip.conf

[general]
register => ServerA:apples@192.168.0.175/serverB
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.177
tcpenable=no
[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia         ; assume device is behind NAT
dtmfmode=auto 
disallow=all
allow=g722
allow=ulaw
allow=alaw

[0000FFFF0001](office-phone)
secret=aaaaVVV2222

[0000FFFF0002](office-phone)
secret=bbbbVVV2222

[ServerB]
type=friend
host=192.168.0.175
port=5060
defaultuser=ServerA
secret=apples
context=incoming
disallow=all
allow = ulaw

extensions.conf

[LocalSets]
exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _6XXX,1,Dial(SIP/ServerB/${EXTEN:1})

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

ServerB: 192.168.0.175

sip.conf

[general]
register => ServerB:apples@192.168.0.177/serverA
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.175
tcpenable=no

[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw 
allow=alaw

[0000FFFF0001](office-phone)
secret=aaaaVVV2222

[0000FFFF0002](office-phone)
secret=bbbbVVV2222

[ServerA]
type=friend
host=192.168.0.177
port=5060
defaultuser=ServerB
secret=apples
context=incoming
disallow=all
allow = ulaw

extensions.conf

[LocalSets]

exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _5XXX,1,Dial(SIP/${EXTEN:1}@ServerA)

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

При наборе на софтфоне с сервера A 6101 получаем: 

    -- Executing [6101@LocalSets:1] Dial("SIP/0000FFFF0001-0000001d", "SIP/ServerB/101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ServerB/101
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/0000FFFF0001-0000001d' status is 'CHANUNAVAIL'

sip show registry

asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
192.168.0.175:5060                      N      ServerA            120 No Authentication
1 SIP registrations.

В общем нужна помощь. Ip tables отключен, звонки на локальные клиенты проходят, никаких натов нет, айпи статические, хотя и висят физически на одной машине в виртуалках, сквозной пинг проходит. Ответ awsswa не помог. zzuz Ладно, хорошо, почему текущий конфиг не работает вообще?вообще? UPDATE Сделал debug для пира, вот что получил 

SIP Debugging Enabled for IP: 192.168.0.175
  == Using SIP RTP CoS mark 5
    -- Executing [6102@LocalSets:1] Dial("SIP/0000FFFF0001-00000002", "SIP/102@ServerB") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14286
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.175:5060:
INVITE sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.1
Date: Thu, 08 May 2014 09:49:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 163586622 163586622 IN IP4 192.168.0.177
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.177
t=0 0
m=audio 14286 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/102@ServerB

<--- SIP read from UDP:192.168.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>;tag=as33978b60
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c72c8df"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.175:5060:
ACK sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>;tag=as33978b60
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0


---
Audio is at 14286
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.175:5060:
INVITE sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username="ServerA", realm="asterisk", algorithm=MD5, uri="sip:102@192.168.0.175", nonce="5c72c8df", response="cf9834a3486aa4aa233a304fbfc8b094"
Date: Thu, 08 May 2014 09:49:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 163586622 163586623 IN IP4 192.168.0.177
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.177
t=0 0
m=audio 14286 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>;tag=as33978b60
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.8.1
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 192.168.0.175:5060:
ACK sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>;tag=as33978b60
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/0000FFFF0001-00000002' status is 'CHANUNAVAIL'
Really destroying SIP dialog '17b2cd570393770d61231afe46bb0981@192.168.0.177:5060' Method: INVITE
Really destroying SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' Method: INVITE

И на втором сервере

<--- SIP read from UDP:192.168.0.177:5060 --->
INVITE sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.1
Date: Thu, 08 May 2014 09:49:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 163586622 163586622 IN IP4 192.168.0.177
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.177
t=0 0
m=audio 14286 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.0.177:5060 (no NAT)
Sending to 192.168.0.177:5060 (no NAT)
Using INVITE request as basis request - 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.0.177:5060

<--- Reliably Transmitting (NAT) to 192.168.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>;tag=as33978b60
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c72c8df"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.177:5060 --->
ACK sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>;tag=as33978b60
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.177:5060 --->
INVITE sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username="ServerA", realm="asterisk", algorithm=MD5, uri="sip:102@192.168.0.175", nonce="5c72c8df", response="cf9834a3486aa4aa233a304fbfc8b094"
Date: Thu, 08 May 2014 09:49:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 163586622 163586623 IN IP4 192.168.0.177
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.177
t=0 0
m=audio 14286 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.0.177:5060 (NAT)
Using INVITE request as basis request - 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.0.177:5060

<--- Reliably Transmitting (NAT) to 192.168.0.177:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>;tag=as33978b60
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.177:5060 --->
ACK sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
To: <sip:102@192.168.0.175>;tag=as33978b60
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '17b2cd570393770d61231afe46bb0981@192.168.0.177:5060' Method: ACK
Really destroying SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' Method: ACK

sip trunk между asterisk

Есть пример из книжки Asterisk: the definitive guide 2013 года. Но не получается. Два однотипных Asterisk 11.8.1 Поставленных с нуля.

ServerA: 192.168.0.177

sip.conf

 [general]
allowguest=no                   ; disable unauthenticated calls
srvlookup=no                    ; disable DNS SRV record lookup on outbound calls
                                ; (unless you have a reliable DNS connection,
                                ; in which case yes)
udpbindaddr=192.168.0.177               ; listen for UDP requests on all interfaces
tcpenable=no                    ; disable TCP support

[office-phone](!)               ; create a template for our devices
type=friend                     ; the channel driver will match on username first,
                                ; IP second
context=LocalSets               ; this is where calls from the device will ente
                                ; the dialplan
host=dynamic                    ; the device will register => ServerA:apples@192.168.0.175/serverB
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.177
tcpenable=no
[office-phone](!)
type=friend
context=LocalSets
host=dynamic
with asterisk
nat=force_rport,comedia         ; assume device is behind NAT
; *** NAT stands for Network Address Translation,
; which allows multiple internal devices to share an
; external IP address.
dtmfmode=auto 
disallow=all
allow=g722
allow=ulaw
                  ; accept touch-tones from the devices, negotiated
                                ; automatically
disallow=all                    ; reset which voice codecs this device will accept or offer
allow=g722                      ; audio codecs to accept from, and request to, the device
allow=ulaw                      ; in the order we prefer
allow=alaw
; define a device name and use the office-phone template

[0000FFFF0001](office-phone)
secret=aaaaVVV2222
secret=aaaaVVV2222 ; a unique password for this device --

[0000FFFF0002](office-phone)
secret=bbbbVVV2222
secret=bbbbVVV2222 ; a unique password for this device --

[ServerB]
type=friend
type=peer
host=192.168.0.175
port=5060
defaultuser=ServerA
secret=apples
secret=111
context=incoming
disallow=all
allow = ulaw
allow=ulaw

extensions.conf

[LocalSets]
exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _6XXX,1,Dial(SIP/ServerB/${EXTEN:1})

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

ServerB: 192.168.0.175

sip.conf

[general]
allowguest=no                   ; disable unauthenticated calls
srvlookup=no                    ; disable DNS SRV record lookup on outbound calls
                                ; (unless you have a reliable DNS connection,
                                ; in which case yes)
udpbindaddr=192.168.0.175               ; listen for UDP requests on all interfaces
tcpenable=no                    ; disable TCP support

[office-phone](!)               ; create a template for our devices
type=friend                     ; the channel driver will match on username first,
                                ; IP second
context=LocalSets               ; this is where calls from the device will ente
                                ; the dialplan
host=dynamic                    ; the device will register => ServerB:apples@192.168.0.177/serverA
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=192.168.0.175
tcpenable=no

[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
with asterisk
nat=force_rport,comedia         ; assume device is behind NAT
; *** NAT stands for Network Address Translation,
; which allows multiple internal devices to share an
; external IP address.
dtmfmode=auto                   ; accept touch-tones from the devices, negotiated
                                ; automatically
disallow=all                    ; reset which voice codecs this device will accept or offer
allow=g722                      ; audio codecs to accept from, and request to, the device
allow=ulaw                       ; in the order we prefer
allow=alaw
; define a device name and use the office-phone template

[0000FFFF0001](office-phone)
secret=aaaaVVV2222
secret=aaaaVVV2222 ; a unique password for this device --

[0000FFFF0002](office-phone)
secret=bbbbVVV2222
secret=bbbbVVV2222 ; a unique password for this device --

[ServerA]
type=friend
type=peer
host=192.168.0.177
port=5060
defaultuser=ServerB
secret=apples
secret=111
context=incoming
disallow=all
allow = ulaw
allow=ulaw

extensions.conf

[LocalSets]

exten => 101,1,Dial(SIP/0000FFFF0001)
exten => 102,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Progress()
 same => n,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()

exten => 201,1,Goto(TestMenu,start,1)
include => external

[external]
exten => _5XXX,1,Dial(SIP/${EXTEN:1}@ServerA)

[TestMenu]
exten => start,1,Answer()
 same => n,Background(enter-ext-of-person)
 same => n,WaitExten(5)

exten => 1,1,Playback(digits/1)
 same => n,Goto(TestMenu,start,1)

exten => 2,1,Playback(digits/2)
 same => n,Goto(TestMenu,start,1)

exten => 0,1,Playback(vm-goodbye)
 same => n,Hangup()

exten => i,1,Playback(pbx-invalid)
 same => n,Goto(TestMenu,start,1)

exten => t,1,Playback(vm-goodbye)
 same => n,Hangup()

[incoming]
exten => _XXX,1,Goto(LocalSets,${EXTEN},1)

При наборе на софтфоне с сервера A 6101 получаем: 

    -- Executing [6101@LocalSets:1] Dial("SIP/0000FFFF0001-0000001d", "SIP/ServerB/101") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ServerB/101
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/0000FFFF0001-0000001d' status is 'CHANUNAVAIL'

sip show registry

asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
192.168.0.175:5060                      N      ServerA            120 No Authentication
1 SIP registrations.

В общем нужна помощь. Ip tables отключен, звонки на локальные клиенты проходят, никаких натов нет, айпи статические, хотя и висят физически на одной машине в виртуалках, сквозной пинг проходит. Ответ awsswa не помог.

zzuz Ладно, хорошо, почему текущий конфиг не работает вообще? вообще?

UPDATE Сделал debug все точно как в книге, включил дебаг для пира, вот что получил пира

SIP Debugging Enabled for IP: 192.168.0.175
  == Using SIP RTP CoS mark 5
    -- Executing [6102@LocalSets:1] Dial("SIP/0000FFFF0001-00000002", Dial("SIP/0000FFFF0001-00000004", "SIP/102@ServerB") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14286
11532
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.175:5060:
INVITE sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa
192.168.0.177:5060;branch=z9hG4bK0e638415
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.1
Date: Thu, 08 May 2014 09:49:12 10:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235
237

v=0
o=root 163586622 163586622 1718078034 1718078034 IN IP4 192.168.0.177
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.177
t=0 0
m=audio 14286 11532 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/102@ServerB

<--- SIP read from UDP:192.168.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa;received=192.168.0.177;rport=5060
192.168.0.177:5060;branch=z9hG4bK0e638415;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as33978b60
<sip:102@192.168.0.175>;tag=as0fdd78d3
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c72c8df"
nonce="68304c97"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.175:5060:
ACK sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa
192.168.0.177:5060;branch=z9hG4bK0e638415
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as33978b60
<sip:102@192.168.0.175>;tag=as0fdd78d3
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0


---
Audio is at 14286
11532
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.175:5060:
INVITE sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee
192.168.0.177:5060;branch=z9hG4bK52b46733
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username="ServerA", realm="asterisk", algorithm=MD5, uri="sip:102@192.168.0.175", nonce="5c72c8df", response="cf9834a3486aa4aa233a304fbfc8b094"
nonce="68304c97", response="de5f5f97eb8aacb9e6705aab96bf148c"
Date: Thu, 08 May 2014 09:49:12 10:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235
237

v=0
o=root 163586622 163586623 1718078034 1718078035 IN IP4 192.168.0.177
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.177
t=0 0
m=audio 14286 11532 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee;received=192.168.0.177;rport=5060
192.168.0.177:5060;branch=z9hG4bK52b46733;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as33978b60
<sip:102@192.168.0.175>;tag=as0fdd78d3
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.8.1
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 192.168.0.175:5060:
ACK sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee
192.168.0.177:5060;branch=z9hG4bK52b46733
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as33978b60
<sip:102@192.168.0.175>;tag=as0fdd78d3
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' '47e3f4892beb1675570863991419e4b9@192.168.0.177:5060' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/0000FFFF0001-00000002' 'SIP/0000FFFF0001-00000004' status is 'CHANUNAVAIL'
Really destroying SIP dialog '17b2cd570393770d61231afe46bb0981@192.168.0.177:5060' Method: INVITE
Really destroying SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' Method: INVITE

И на втором сервересервер B:

<--- SIP read from UDP:192.168.0.177:5060 --->
INVITE sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa
192.168.0.177:5060;branch=z9hG4bK0e638415
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.1
Date: Thu, 08 May 2014 09:49:12 10:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235
237

v=0
o=root 163586622 163586622 1718078034 1718078034 IN IP4 192.168.0.177
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.177
t=0 0
m=audio 14286 11532 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.0.177:5060 (no NAT)
Sending to 192.168.0.177:5060 (no NAT)
Using INVITE request as basis request - 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.0.177:5060

<--- Reliably Transmitting (NAT) to 192.168.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa;received=192.168.0.177;rport=5060
192.168.0.177:5060;branch=z9hG4bK0e638415;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as33978b60
<sip:102@192.168.0.175>;tag=as0fdd78d3
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c72c8df"
nonce="68304c97"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' '47e3f4892beb1675570863991419e4b9@192.168.0.177:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.177:5060 --->
ACK sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK1da4dfaa
192.168.0.177:5060;branch=z9hG4bK0e638415
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as33978b60
<sip:102@192.168.0.175>;tag=as0fdd78d3
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.177:5060 --->
INVITE sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee
192.168.0.177:5060;branch=z9hG4bK52b46733
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username="ServerA", realm="asterisk", algorithm=MD5, uri="sip:102@192.168.0.175", nonce="5c72c8df", response="cf9834a3486aa4aa233a304fbfc8b094"
nonce="68304c97", response="de5f5f97eb8aacb9e6705aab96bf148c"
Date: Thu, 08 May 2014 09:49:12 10:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235
237

v=0
o=root 163586622 163586623 1718078034 1718078035 IN IP4 192.168.0.177
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.177
t=0 0
m=audio 14286 11532 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.0.177:5060 (NAT)
Using INVITE request as basis request - 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.0.177:5060

<--- Reliably Transmitting (NAT) to 192.168.0.177:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee;received=192.168.0.177;rport=5060
192.168.0.177:5060;branch=z9hG4bK52b46733;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as33978b60
<sip:102@192.168.0.175>;tag=as0fdd78d3
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' '47e3f4892beb1675570863991419e4b9@192.168.0.177:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.177:5060 --->
ACK sip:102@192.168.0.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.177:5060;branch=z9hG4bK26472eee
192.168.0.177:5060;branch=z9hG4bK52b46733
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as470b2f8f
<sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as33978b60
<sip:102@192.168.0.175>;tag=as0fdd78d3
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 007a7e156524de362cb0193b22c64a24@192.168.0.177:5060
47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '17b2cd570393770d61231afe46bb0981@192.168.0.177:5060' Method: ACK
Really destroying SIP dialog '007a7e156524de362cb0193b22c64a24@192.168.0.177:5060' '47e3f4892beb1675570863991419e4b9@192.168.0.177:5060' Method: ACK
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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.