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 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 ; a unique password for this device --
[0000FFFF0002](office-phone)
secret=bbbbVVV2222 ; a unique password for this device --
[ServerB]
type=peer
host=192.168.0.175
defaultuser=ServerA
secret=111
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]
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 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 ; a unique password for this device --
[0000FFFF0002](office-phone)
secret=bbbbVVV2222 ; a unique password for this device --
[ServerA]
type=peer
host=192.168.0.177
defaultuser=ServerB
secret=111
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'
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?
UPDATE Сделал все точно как в книге, включил дебаг для пира
SIP Debugging Enabled for IP: 192.168.0.175
== Using SIP RTP CoS mark 5
-- Executing [6102@LocalSets:1] Dial("SIP/0000FFFF0001-00000004", "SIP/102@ServerB") in new stack
== Using SIP RTP CoS mark 5
Audio is at 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=z9hG4bK0e638415
Max-Forwards: 70
From: <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: 47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.1
Date: Thu, 08 May 2014 10:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 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 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=z9hG4bK0e638415;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as0fdd78d3
Call-ID: 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="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=z9hG4bK0e638415
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as0fdd78d3
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0
---
Audio is at 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=z9hG4bK52b46733
Max-Forwards: 70
From: <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: 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="68304c97", response="de5f5f97eb8aacb9e6705aab96bf148c"
Date: Thu, 08 May 2014 10:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 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 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=z9hG4bK52b46733;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as0fdd78d3
Call-ID: 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=z9hG4bK52b46733
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as0fdd78d3
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0
---
Scheduling destruction of SIP dialog '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/0000FFFF0001-00000004' status is 'CHANUNAVAIL'
И сервер 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=z9hG4bK0e638415
Max-Forwards: 70
From: <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: 47e3f4892beb1675570863991419e4b9@192.168.0.177:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.1
Date: Thu, 08 May 2014 10:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 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 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 - 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=z9hG4bK0e638415;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as0fdd78d3
Call-ID: 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="68304c97"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '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=z9hG4bK0e638415
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as0fdd78d3
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 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=z9hG4bK52b46733
Max-Forwards: 70
From: <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: 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="68304c97", response="de5f5f97eb8aacb9e6705aab96bf148c"
Date: Thu, 08 May 2014 10:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 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 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 - 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=z9hG4bK52b46733;received=192.168.0.177;rport=5060
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as0fdd78d3
Call-ID: 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 '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=z9hG4bK52b46733
Max-Forwards: 70
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>;tag=as0fdd78d3
Contact: <sip:0000FFFF0001@192.168.0.177:5060>
Call-ID: 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 '47e3f4892beb1675570863991419e4b9@192.168.0.177:5060' Method: ACK
У Вас явно прописаны IP.. Может Вам и не нужна авторизация?
[serverA]
type = friend
host = 192.168.1.101
context = incoming
disallow = all
allow = ulaw
[ServerB]
type=friend
host=192.168.0.175
context=incoming
disallow=all
allow=ulaw
From: <sip:0000FFFF0001@192.168.0.177>;tag=as725d5ac3
To: <sip:102@192.168.0.175>
Это говорит о том, что у пиров не настроено поле fromuser=
На будущее, читайте правильные источники..
Там же четко расписано:
[mysipprovider-out]
type=peer
secret=password
username=2345
host=sipserver.mysipprovider.com
fromuser=2345
fromdomain=fwd.pulver.com
canreinvite=no
insecure=invite,port
qualify=yes
nat=yes
context=from-mysipprovider ; this section will be defined in extensions.conf
Чтобы что-то регистрировать на другом сервере на нем должно быть выставлено host=dynamic . Внимательней читайте книгу.
Не работает , потому что серверу B посылаетеся неаунтентифицированный INVITE . Во-первых для регистрации в настройках пира нет ни логина , ни пароля. во-вторых, если их не использовать то необходим insecure=invite для входящего вызова. Выдирать часть конфига из книги и потом разводить руками - неверный путь.
zzuz (May 8 '14)editЗадан: May 8 '14
Просмотрен: 5,108 раз
Обновлен: May 09 '14
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.
Посто обновлен
tw1911 (May 8 '14)editПосле всех ответов рабочий конфиг так и не выработан. Добавление insecure=invite не дает желанного результата. Еще раз поясню, что уменно такой конфиг был в книге, я его не вырывал и принципиально не менял.
tw1911 (May 8 '14)editВы вызываете ServerB/num , в котором defaultuser=ServerA , но в конфиге ServerA стоит defaultuser=ServerB .
zzuz (May 8 '14)edit