История изменений [назад]

В общем опять возобновилось тестирование. Все то же самое по настройкам trixbox:

PEER Details: disallow=all context=default allow=ulaw,alaw,g729 bindaddr=0.0.0.0 bindport=5060 srvlookup=yes echocancel=no dtmfmode=rfc2833 qualify=yes insecure=invite,port

User Details: reinvite=yes host=188.254.68.76&188.254.69.76 port=5060 insecure=invite,port type=friend

Так вот при звонке на номер 8 800ххххх астериск пишет: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 -- Executing [5555366703@from-sip-external:1] NoOp("SIP/sipgw120.com-0000203d", "Received incoming SIP connection from unknown peer to 5555366703") in new stack -- Executing [5555366703@from-sip-external:2] Set("SIP/sipgw120.com-0000203d", "DID=5555366703") in new stack -- Executing [5555366703@from-sip-external:3] Goto("SIP/sipgw120.com-0000203d", "s,1") in new stack -- Goto (from-sip-external,s,1) -- Executing [s@from-sip-external:1] GotoIf("SIP/sipgw120.com-0000203d", "0?from-trunk,5555366703,1") in new stack -- Executing [s@from-sip-external:2] Set("SIP/sipgw120.com-0000203d", "TIMEOUT(absolute)=15") in new stack Channel will hangup at 2012-05-22 12:51:26.000 YEKST. -- Executing [s@from-sip-external:3] Answer("SIP/sipgw120.com-0000203d", "") in new stack -- Executing [s@from-sip-external:4] Wait("SIP/sipgw120.com-0000203d", "2") in new stack -- Executing [s@from-sip-external:5] Playback("SIP/sipgw120.com-0000203d", "ss-noservice") in new stack -- <sip sipgw120.com-0000203d=""> Playing 'ss-noservice.gsm' (language 'en') -- Executing [s@from-sip-external:6] PlayTones("SIP/sipgw120.com-0000203d", "congestion") in new stack -- Executing [s@from-sip-external:7] Congestion("SIP/sipgw120.com-0000203d", "5") in new stack == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/sipgw120.com-0000203d' -- Executing [h@from-sip-external:1] NoOp("SIP/sipgw120.com-0000203d", "Hangup") in new stack -- Executing [h@from-sip-external:2] Set("SIP/sipgw120.com-0000203d", "DID=s") in new stack -- Executing [h@from-sip-external:3] Goto("SIP/sipgw120.com-0000203d", "s,1") in new stack -- Goto (from-sip-external,s,1) -- Executing [s@from-sip-external:1] GotoIf("SIP/sipgw120.com-0000203d", "0?from-trunk,s,1") in new stack -- Executing [s@from-sip-external:2] Set("SIP/sipgw120.com-0000203d", "TIMEOUT(absolute)=15") in new stack Channel will hangup at 2012-05-22 12:51:39.000 YEKST. -- Executing [s@from-sip-external:3] Answer("SIP/sipgw120.com-0000203d", "") in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/sipgw120.com-0000203d'

и в трубке тишина ( т.е. должен IVR бубнить .. на него направил incoming). Говорят что просто голос на астериске включить надо. Порты все открыты в том числе 5060 и с 10000по 65000 UDP. Что делаю не так ?

В общем опять возобновилось тестирование. Все то же самое по настройкам trixbox:

PEER Details:
disallow=all
context=default
allow=ulaw,alaw,g729
bindaddr=0.0.0.0
bindport=5060
srvlookup=yes
echocancel=no
dtmfmode=rfc2833
qualify=yes
insecure=invite,port
 
insecure=invite,port

User Details:
reinvite=yes
host=188.254.68.76&188.254.69.76
port=5060
insecure=invite,port
type=friend
 
type=friend


Так вот при звонке на номер 8 800ххххх 
астериск пишет:
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
    -- Executing [5555366703@from-sip-external:1] NoOp("SIP/sipgw120.com-0000203d", "Received incoming SIP connection from unknown peer to 5555366703") in new stack
    -- Executing [5555366703@from-sip-external:2] Set("SIP/sipgw120.com-0000203d", "DID=5555366703") in new stack
    -- Executing [5555366703@from-sip-external:3] Goto("SIP/sipgw120.com-0000203d", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/sipgw120.com-0000203d", "0?from-trunk,5555366703,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/sipgw120.com-0000203d", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2012-05-22 12:51:26.000 YEKST.
    -- Executing [s@from-sip-external:3] Answer("SIP/sipgw120.com-0000203d", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/sipgw120.com-0000203d", "2") in new stack
    -- Executing [s@from-sip-external:5] Playback("SIP/sipgw120.com-0000203d", "ss-noservice") in new stack
    -- <sip sipgw120.com-0000203d=""> <SIP/sipgw120.com-0000203d> Playing 'ss-noservice.gsm' (language 'en')
    -- Executing [s@from-sip-external:6] PlayTones("SIP/sipgw120.com-0000203d", "congestion") in new stack
    -- Executing [s@from-sip-external:7] Congestion("SIP/sipgw120.com-0000203d", "5") in new stack
  == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/sipgw120.com-0000203d'
    -- Executing [h@from-sip-external:1] NoOp("SIP/sipgw120.com-0000203d", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/sipgw120.com-0000203d", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/sipgw120.com-0000203d", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/sipgw120.com-0000203d", "0?from-trunk,s,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/sipgw120.com-0000203d", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2012-05-22 12:51:39.000 YEKST.
    -- Executing [s@from-sip-external:3] Answer("SIP/sipgw120.com-0000203d", "") in new stack
  == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/sipgw120.com-0000203d''SIP/sipgw120.com-0000203d'