История изменений [назад]

Привет. При входящем вызове у меня в логах появляется

[Jun 14 12:10:10] VERBOSE[22998] config.c:   == Parsing '/etc/asterisk/logger.conf': [Jun 14 12:10:10] VERBOSE[22998] config.c:   == Found
[Jun 14 12:10:10] VERBOSE[22998] logger.c:  Asterisk Queue Logger restarted
[Jun 14 12:10:18] VERBOSE[21917] netsock2.c:   == Using SIP RTP TOS bits 184
[Jun 14 12:10:18] VERBOSE[21917] netsock2.c:   == Using SIP RTP CoS mark 5
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:1] GotoIf("SIP/217.15.55.203:5061-0000003b", "0?checklang:noanonymous") in new stack
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Goto (from-sip-external,s,5)
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:5] Set("SIP/217.15.55.203:5061-0000003b", "TIMEOUT(absolute)=15") in new stack
[Jun 14 12:10:18] VERBOSE[22999] func_timeout.c: Channel will hangup at 2012-06-14 12:10:33.975 MSD.
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:6] Answer("SIP/217.15.55.203:5061-0000003b", "") in new stack
[Jun 14 12:10:19] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:7] Wait("SIP/217.15.55.203:5061-0000003b", "2") in new stack
[Jun 14 12:10:21] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:8] Playback("SIP/217.15.55.203:5061-0000003b", "ss-noservice") in new stack
[Jun 14 12:10:21] VERBOSE[22999] file.c:     -- <SIP/217.15.55.203:5061-0000003b> Playing 'ss-noservice.gsm' (language 'en')
[Jun 14 12:10:26] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:9] PlayTones("SIP/217.15.55.203:5061-0000003b", "congestion") in new stack
[Jun 14 12:10:26] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:10] Congestion("SIP/217.15.55.203:5061-0000003b", "5") in new stack
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:   == Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/217.15.55.203:5061-0000003b'
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:     -- Executing [h@from-sip-external:1] Hangup("SIP/217.15.55.203:5061-0000003b", "") in new stack
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:   == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/217.15.55.203:5061-0000003b'
[Jun 14 12:10:29] VERBOSE[21917] dnsmgr.c:        > doing dnsmgr_lookup for 'sipnet.ru'
[Jun 14 12:10:32] VERBOSE[22998] asterisk.c:     -- Remote UNIX connection disconnected

Что за IP такой 217.15.55.203, без понятия. Точно не наш. Транк настроен как обычно

dtmfmode=rfc2833
type=friend
host=217.15.55.200
fromuser=6XXX
username=495YYYYYYY
secret=Secr
insecure=port,invite
conext=contex-internal
nat=no
allow=all

Строка регистрации 6XXX:secr:495YYYYYYY@217.15.55.20 Входящая маршрутизация на внутренний номер.

И еще замечена странная штука, при звонке на наш номер с мобильного, когда астериск начинает говорить что номер не существует, на экране мобильного номер меняется на другой 4959269211, т.е не на тот на который звоним. FreePBX 2.8, Centos 5.7 Спасибо.

Привет. При входящем вызове у меня в логах появляется

[Jun 14 12:10:10] VERBOSE[22998] config.c:   == Parsing '/etc/asterisk/logger.conf': [Jun 14 12:10:10] VERBOSE[22998] config.c:   == Found
[Jun 14 12:10:10] VERBOSE[22998] logger.c:  Asterisk Queue Logger restarted
[Jun 14 12:10:18] VERBOSE[21917] netsock2.c:   == Using SIP RTP TOS bits 184
[Jun 14 12:10:18] VERBOSE[21917] netsock2.c:   == Using SIP RTP CoS mark 5
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:1] GotoIf("SIP/217.15.55.203:5061-0000003b", "0?checklang:noanonymous") in new stack
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Goto (from-sip-external,s,5)
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:5] Set("SIP/217.15.55.203:5061-0000003b", "TIMEOUT(absolute)=15") in new stack
[Jun 14 12:10:18] VERBOSE[22999] func_timeout.c: Channel will hangup at 2012-06-14 12:10:33.975 MSD.
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:6] Answer("SIP/217.15.55.203:5061-0000003b", "") in new stack
[Jun 14 12:10:19] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:7] Wait("SIP/217.15.55.203:5061-0000003b", "2") in new stack
[Jun 14 12:10:21] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:8] Playback("SIP/217.15.55.203:5061-0000003b", "ss-noservice") in new stack
[Jun 14 12:10:21] VERBOSE[22999] file.c:     -- <SIP/217.15.55.203:5061-0000003b> Playing 'ss-noservice.gsm' (language 'en')
[Jun 14 12:10:26] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:9] PlayTones("SIP/217.15.55.203:5061-0000003b", "congestion") in new stack
[Jun 14 12:10:26] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:10] Congestion("SIP/217.15.55.203:5061-0000003b", "5") in new stack
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:   == Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/217.15.55.203:5061-0000003b'
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:     -- Executing [h@from-sip-external:1] Hangup("SIP/217.15.55.203:5061-0000003b", "") in new stack
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:   == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/217.15.55.203:5061-0000003b'
[Jun 14 12:10:29] VERBOSE[21917] dnsmgr.c:        > doing dnsmgr_lookup for 'sipnet.ru'
[Jun 14 12:10:32] VERBOSE[22998] asterisk.c:     -- Remote UNIX connection disconnected

Что за IP такой 217.15.55.203, без понятия. Точно не наш. Транк настроен как обычно

dtmfmode=rfc2833
type=friend
host=217.15.55.200
fromuser=6XXX
username=495YYYYYYY
secret=Secr
insecure=port,invite
conext=contex-internal
nat=no
allow=all

Строка регистрации 6XXX:secr:495YYYYYYY@217.15.55.20 6XXX:secr:495YYYYYYY@217.15.55.200 Входящая маршрутизация на внутренний номер.

И еще замечена странная штука, при звонке на наш номер с мобильного, когда астериск начинает говорить что номер не существует, на экране мобильного номер меняется на другой 4959269211, т.е не на тот на который звоним. FreePBX 2.8, Centos 5.7 Спасибо.