asterisk.ru

Русский форум Asterisk сообщества
https://forum.asterisk.ru/

Asterisk не хочет отдавать SIP 484

https://forum.asterisk.ru/viewtopic.php?t=512

Страница 1 из 1

Asterisk не хочет отдавать SIP 484

Добавлено: 05 май 2011, 14:32
gladov
Добрый день.

Пытаюсь настроить Asterisk 1.6.2.9 (debian rep) + Grandstream FXS шлюзы. У этих самых Grandsteram есть функция Early Dial, т.е. они могут слать SIP INVITE после каждой цифры, а asterisk должен отвечать SIP 484 если данного номера нет в номерном плане. Так вот он после первой же цифры вместо 484 дает полный отлуп в виде 404:

Код: Выделить всё

[May  5 14:26:39] DEBUG[21191] acl.c: Found IP address for this socket
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.1:5060
[May  5 14:26:39] VERBOSE[21191] netsock.c:   == Using SIP RTP CoS mark 5
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Setting NAT on RTP to Off
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Allocating new SIP dialog for 587137444-5062-5@192.168.1.98 - INVITE (With RTP)
[May  5 14:26:39] DEBUG[21191] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Setting NAT on RTP to Off
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.98:5062
[May  5 14:26:39] DEBUG[21191] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Stopping retransmission on '587137444-5062-5@192.168.1.98' of Response 40: Match Found
[May  5 14:26:39] DEBUG[21191] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Setting NAT on RTP to Off
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing session-level SDP o=111 8002 8000 IN IP4 192.168.1.98... UNSUPPORTED.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.98... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=fmtp:97 mode=20... UNSUPPORTED.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 G729E/8000... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:100 AAL2-G726-16/8000... OK.
[May  5 14:26:39] DEBUG[21191] chan_sip.c: We're settling with these formats: 0x4 (ulaw)
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Checking SIP call limits for device 111
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Updating call counter for incoming call
[May  5 14:26:39] DEBUG[21179] devicestate.c: No provider found, checking channel drivers for SIP - 111
[May  5 14:26:39] DEBUG[21179] chan_sip.c: Checking device state for peer 111
[May  5 14:26:39] DEBUG[21191] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 192.168.1.98:5062
[May  5 14:26:39] DEBUG[21179] devicestate.c: Changing state for SIP/111 - state 2 (In use)
[May  5 14:26:39] DEBUG[21179] devicestate.c: device 'SIP/111' state '2'
[May  5 14:26:39] NOTICE[21191] chan_sip.c: Call from '111' to extension '1' rejected because extension not found in context 'DLPN_DialPlan1'.
Вот весь dialplan:

Код: Выделить всё

PowerEdge*CLI> dialplan show
[ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ]
  's' =>            1. NoOp()                                     [app_dial]

[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
  's' =>            1. NoOp()                                     [app_queue]

[ Context 'parkedcalls' created by 'features' ]
  '700' =>          1. Park()                                     [features]

[ Context 'demo' created by 'pbx_lua' ]
  Alt. Switch =>    'Lua/'                                        [pbx_lua]

[ Context 'ringroups-custom-3' created by 'pbx_config' ]
  's' =>            1. NoOp(Buh_all)                              [pbx_config]
                    2. Dial(SIP/101&SIP/108&SIP/109,60,${DIALOPTIONS}i) [pbx_config]
                    3. Hangup()                                   [pbx_config]

[ Context 'ringroups-custom-2' created by 'pbx_config' ]
  's' =>            1. NoOp(Buh)                                  [pbx_config]
                    2. Dial(SIP/109,10,${DIALOPTIONS}i)           [pbx_config]
                    3. Dial(SIP/108,10,${DIALOPTIONS}i)           [pbx_config]
                    4. Dial(SIP/101,10,${DIALOPTIONS}i)           [pbx_config]
                    5. Goto(ringroups-custom-3,s,1)               [pbx_config]

[ Context 'ringroups-custom-1' created by 'pbx_config' ]
  's' =>            1. NoOp(Managers)                             [pbx_config]
                    2. Dial(SIP/104,10,${DIALOPTIONS}i)           [pbx_config]
                    3. Dial(SIP/107,10,${DIALOPTIONS}i)           [pbx_config]
                    4. Dial(SIP/102,10,${DIALOPTIONS}i)           [pbx_config]
                    5. Goto(ringroups-custom-4,s,1)               [pbx_config]

[ Context 'ringroups-custom-4' created by 'pbx_config' ]
  's' =>            1. NoOp(Managers_all)                         [pbx_config]
                    2. Dial(SIP/102&SIP/104&SIP/107,60,${DIALOPTIONS}i) [pbx_config]
                    3. Hangup()                                   [pbx_config]

[ Context 'DID_trunk_1' created by 'pbx_config' ]
  's' =>            1. Goto(ringroups-custom-1,s,1)               [pbx_config]

[ Context 'outgoing' created by 'pbx_config' ]
  '_9X.' =>         1. Dial(SIP/trunk_1/${EXTEN:1},,tT)           [pbx_config]
                    2. Congestion()                               [pbx_config]
                    3. Hangup()                                   [pbx_config]

[ Context 'DLPN_DialPlan1' created by 'pbx_config' ]
  Include =>        'default'                                     [pbx_config]
  Include =>        'outgoing'                                    [pbx_config]

[ Context 'ringgroups' created by 'pbx_config' ]
  '220' =>          1. Goto(ringroups-custom-1,s,1)               [pbx_config]
  '221' =>          1. Goto(ringroups-custom-2,s,1)               [pbx_config]
  '222' =>          1. Goto(ringroups-custom-3,s,1)               [pbx_config]
  '223' =>          1. Goto(ringroups-custom-4,s,1)               [pbx_config]

[ Context 'default' created by 'pbx_config' ]
  '101' =>          hint: SIP/101                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '102' =>          hint: SIP/102                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '103' =>          hint: SIP/103                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '104' =>          hint: SIP/104                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '105' =>          hint: SIP/105                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '106' =>          hint: SIP/106                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '107' =>          hint: SIP/107                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '108' =>          hint: SIP/108                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '109' =>          hint: SIP/109                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '110' =>          hint: SIP/110                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '111' =>          hint: SIP/111                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '112' =>          hint: SIP/112                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '113' =>          hint: SIP/113                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '199' =>          hint: SIP/199                                 [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  Include =>        'DID_trunk_1'                                 [pbx_config]

-= 27 extensions (55 priorities) in 13 contexts. =-
Как можно заставить его отправлять 484 ответ?
Часовой пояс: UTC+03:00
Страница 1 из 1

Создано на основе phpBB® Forum Software © phpBB Limited

Русская поддержка phpBB