десятки страниц прочитал, просмотрел, все делал как написано - НО НЕ ХОЧЕТ зараза звонить ! (баланс пополнил, оператор сипнет)
исходящая маршрутизация:
8XXXXXXXXXXX
XXXXXXX
Транк:
7351+XXXXXXX
7+8|XXXXXXXXXX
Пир:
dtmfmode=rfc2833
type=friend
host=212.53.40.40
fromuser=0024009240
fromdomain=212.53.40.40
secret=Пароль
username=0024009240
insecure=port,invite
canreinvite=no
conext=contex-internal
disallow=all
nat=yes
allow=g729&g723&g723.1&gsm&ulaw&alaw
Outbound Dial Prefix: 9
context=from-trunk
secret=Пароль
type=user
регистрация:
0024009240:Пароль@212.53.40.40/0024009240
Регистрируется нормально.
набираю так: 97359077, 99517733111 и так пробовал 989517733111
Added after 4 minutes:
Логи:
уже весь мозг взорвал
09:07:58 SIP.Registration INFO Scheduling NAT/Firewall binding refresh for sip:112@192.168.1.20 after 20 seconds
09:08:18 SIP.Registration INFO NAT/Firewall binding refresh by dummy packet to UDP:192.168.1.20:5060
09:08:18 SIP.Network DEBUG
2009-12-16 04:08:18.705 UDP LOCAL->192.168.1.20:5060
09:08:18 SIP.Registration INFO Scheduling NAT/Firewall binding refresh for sip:112@192.168.1.20 after 20 seconds
09:08:25 SIPCall DEBUG SIP Call 38: Initiate to sip:97359077@192.168.1.20
09:08:25 SIP.Network DEBUG
2009-12-16 04:08:25.940 UDP LOCAL->192.168.1.20:5060
INVITE sip:97359077@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002b4b285d39000015fd00000010;rport
From: "unknown" ;tag=4cf7a70f7d8
To:
Contact:
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 368
Content-Type: application/sdp
Supported: replaces,norefersub,timer
v=0
o=- 3469925305 3469925305 IN IP4 192.168.1.111
s=SJphone
c=IN IP4 192.168.1.111
t=0 0
m=audio 49156 RTP/AVP 3 97 98 8 0 101
c=IN IP4 192.168.1.111
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv
09:08:26 SIP.Network DEBUG
2009-12-16 04:08:26.158 UDP 192.168.1.20:5060->LOCAL
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002b4b285d39000015fd00000010;received=192.168.1.111;rport=49152
From: "unknown" ;tag=4cf7a70f7d8
To: ;tag=as653a22a3
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a980fc8"
Content-Length: 0
09:08:26 SIP.Network DEBUG
2009-12-16 04:08:26.158 UDP LOCAL->192.168.1.20:5060
ACK sip:97359077@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002b4b285d39000015fd00000010;rport
From: "unknown" ;tag=4cf7a70f7d8
To: ;tag=as653a22a3
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
09:08:26 SIP.Network DEBUG
2009-12-16 04:08:26.158 UDP LOCAL->192.168.1.20:5060
INVITE sip:97359077@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002c4b285d3a00006e1900000012;rport
From: "unknown" ;tag=4cf7a70f7d8
To:
Contact:
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 2 INVITE
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 368
Content-Type: application/sdp
Supported: replaces,norefersub,timer
Proxy-Authorization: Digest username="112",realm="asterisk",nonce="4a980fc8",uri="sip:97359077@192.168.1.20",response="bd49d6b7d12336c1d50e161fcc3fdc22",algorithm=MD5
v=0
o=- 3469925305 3469925305 IN IP4 192.168.1.111
s=SJphone
c=IN IP4 192.168.1.111
t=0 0
m=audio 49156 RTP/AVP 3 97 98 8 0 101
c=IN IP4 192.168.1.111
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv
09:08:26 SIP.Network DEBUG
2009-12-16 04:08:26.158 UDP 192.168.1.20:5060->LOCAL
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002c4b285d3a00006e1900000012;received=192.168.1.111;rport=49152
From: "unknown" ;tag=4cf7a70f7d8
To:
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
09:08:27 SIP.Network DEBUG
2009-12-16 04:08:27.174 UDP 192.168.1.20:5060->LOCAL
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002c4b285d3a00006e1900000012;received=192.168.1.111;rport=49152
From: "unknown" ;tag=4cf7a70f7d8
To: ;tag=as2ed0cb0e
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 3096 3096 IN IP4 192.168.1.20
s=session
c=IN IP4 192.168.1.20
t=0 0
m=audio 19928 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
09:08:27 SIPCall INFO SDP Processor dump:
SDP Header:
Processor: "-", ID = 13
Connection: 192.168.1.111
Session:3469925305
Version:3469925305
Open Internet mode: no
Media slot #0: [audio] -- [normal]
Remote:
RTP: 192.168.1.20 : 19928
RTCP: 192.168.1.20 : 19929
Capability: PCMU
RFC2833: enabled [pt:101]
Remote sdp:
m=audio 19928 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
09:08:27 SIPCall INFO Multimedia Session dump:
Multimedia Session dump:
Session ID: 13
IP address: 0.0.0.0:0
2 channels created:
RTP Audio Inbound channel dump:
Status: started
Capability: undefined
Local RTP = 0.0.0.0 : 49156
Local RTCP = 0.0.0.0 : 49157
Remote RTP = 192.168.1.20 : 19928
Remote RTCP = 192.168.1.20 : 19929
RTP Audio Outbound channel dump:
Status: started
Capability: Microsoft CCITT G.711 u-Law CODEC ( Payload Type = 0 )
Local RTP = 0.0.0.0 : 49156
Local RTCP = 0.0.0.0 : 49157
Remote RTP = 192.168.1.20 : 19928
Remote RTCP = 192.168.1.20 : 19929
RFC2833 Enabled ( Payload Type = 101 )
09:08:29 App INFO HangUp
09:08:29 SIPCall DEBUG SIP Call 38 (sip:97359077@192.168.1.20): Hangup (instant)
09:08:29 SIP.Network DEBUG
2009-12-16 04:08:29.408 UDP LOCAL->192.168.1.20:5060
CANCEL sip:97359077@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002c4b285d3a00006e1900000012;rport
From: "unknown" ;tag=4cf7a70f7d8
To:
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
09:08:29 SIP.Network DEBUG
2009-12-16 04:08:29.408 UDP 192.168.1.20:5060->LOCAL
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002c4b285d3a00006e1900000012;received=192.168.1.111;rport=49152
From: "unknown" ;tag=4cf7a70f7d8
To: ;tag=as2ed0cb0e
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
09:08:29 SIP.Network DEBUG
2009-12-16 04:08:29.408 UDP 192.168.1.20:5060->LOCAL
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002c4b285d3a00006e1900000012;received=192.168.1.111;rport=49152
From: "unknown" ;tag=4cf7a70f7d8
To: ;tag=as2ed0cb0e
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
09:08:29 SIP.Network DEBUG
2009-12-16 04:08:29.424 UDP LOCAL->192.168.1.20:5060
ACK sip:97359077@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:49152;branch=z9hG4bKc0a8016f0000002c4b285d3a00006e1900000012;rport
From: "unknown" ;tag=4cf7a70f7d8
To: ;tag=as2ed0cb0e
Call-ID: CA614BF2C42F4D3399F245DC201FD5AF0xc0a8016f
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
Proxy-Authorization: Digest username="112",realm="asterisk",nonce="4a980fc8",uri="sip:97359077@192.168.1.20",response="bd49d6b7d12336c1d50e161fcc3fdc22",algorithm=MD5
09:08:29 App INFO Call 13 ended
09:08:38 SIP.Registration INFO NAT/Firewall binding refresh by dummy packet to UDP:192.168.1.20:5060
09:08:38 SIP.Network DEBUG
2009-12-16 04:08:38.705 UDP LOCAL->192.168.1.20:5060
| vip74 писал(а): |
| conext=contex-internal |
Уже не у первого такое вижу. Может все-таки context?
| zlat писал(а): | ||
| vip74 писал(а): |
| conext=contex-internal |
Уже не у первого такое вижу. Может все-таки context?
согласен, Contex - это другое)))
сам видел, но я копировал, думал так надо
поменял на: context=context-internal
не помогло
у вас прописано на 8+десять цифр или же просто 7 цифр, а набираете вы все с девятки, куда ж он пойдет?
| zlat писал(а): |
| смотрите исх маршрутизацию у вас прописано на 8+десять цифр или же просто 7 цифр, а набираете вы все с девятки, куда ж он пойдет? |
у меня несколько транков. Сипнет - префикс:9, Телфин - префикс:0.
И из-за этого я набраю: 9 (выход на Сипнет) потом 7знач.номер
я уже неделю бьюсь!!! прошу помощи
удалил Телфина, оставил только Сипнет и убрал префикс, все-равно не работает, куда еще копать?
[ Context 'outrt-001-sipnet_out' created by 'pbx_config' ]
'_8XXXXXXXXXXX' => 1. Macro(user-callerid|SKIPTTL|) [pbx_config]
2. Set(_NODEST=) [pbx_config]
3. Macro(record-enable|${AMPUSER}|OUT|) [pbx_config]
4. Macro(dialout-trunk|2|${EXTEN}||) [pbx_config]
5. Macro(outisbusy|) [pbx_config]
'_XXXXXXX' => 1. Macro(user-callerid|SKIPTTL|) [pbx_config]
2. Set(_NODEST=) [pbx_config]
3. Macro(record-enable|${AMPUSER}|OUT|) [pbx_config]
4. Macro(dialout-trunk|2|${EXTEN}||) [pbx_config]
5. Macro(outisbusy|) [pbx_config]
Include => 'outrt-001-sipnet_out-custom' [pbx_config]
[ Context 'outbound-allroutes' created by 'pbx_config' ]
'foo' => 1. Noop(bar) [pbx_config]
Include => 'outbound-allroutes-custom' [pbx_config]
Include => 'outrt-001-sipnet_out' [pbx_config]
Added after 3 hours 59 minutes:
DED помоги хоть ты???
Спасибо Zlat
Правильные настройки Транка:
host=sipnet.ru
fromuser=Sip_Id
fromdomain=sipnet.ru
secret=Пароль
username=Sip_Id
;insecure=port,invite
canreinvite=no
;conetext=from-trunk
type=peer
disallow=all
allow=alaw&ulaw
nat=no
qualify=yes