[asterisk-users] Asterisk strange behaviour

[Stéphane Kamga]

Hi all

 

I’m a newbie to asterisk and I have install and configure asterisk 1.4.5

I have made some test and have face a strange behaviour

 

I hava a simple dialplan when a call is receive from PTSN,

[PSTN]

exten => s,1,Answer()

exten => s,2,Playback(intro-sicx) ; Listen to your voice

exten => s,3,Dial(SIP/steph)

exten => s,4,Hangup()

 

I got the following when a call is issue

   -- Starting simple switch on 'Zap/1-1'

[Jun 30 13:13:17] ERROR[3107]: callerid.c:564 callerid_feed: fsk_serie made
mylen < 0 (-3)

[Jun 30 13:13:17] WARNING[3107]: chan_zap.c:6396 ss_thread: CallerID feed
failed: Success

[Jun 30 13:13:17] WARNING[3107]: chan_zap.c:6496 ss_thread: CallerID
returned with error on channel 'Zap/1-1'

    -- Executing [s at PSTN:1] Answer("Zap/1-1", "") in new stack

    -- Executing [s at PSTN:2] Playback("Zap/1-1", "intro-sicx") in new stack

    -- <Zap/1-1> Playing 'intro-sicx' (language 'en')

    -- Executing [s at PSTN:3] Dial("Zap/1-1", "SIP/steph") in new stack

    -- Called steph

    -- SIP/steph-081d9058 is ringing

    -- SIP/steph-081d9058 is making progress passing it to Zap/1-1

    -- SIP/steph-081d9058 answered Zap/1-1

  == Spawn extension (PSTN, s, 3) exited non-zero on 'Zap/1-1'

    -- Hungup 'Zap/1-1'

    -- Starting simple switch on 'Zap/1-1'

[Jun 30 13:15:31] WARNING[3119]: chan_zap.c:6496 ss_thread: CallerID
returned with error on channel 'Zap/1-1'

    -- Executing [s at PSTN:1] Answer("Zap/1-1", "") in new stack

    -- Executing [s at PSTN:2] Playback("Zap/1-1", "intro-sicx") in new stack

    -- <Zap/1-1> Playing 'intro-sicx' (language 'en')

    -- Executing [s at PSTN:3] Dial("Zap/1-1", "SIP/steph") in new stack

    -- Called steph

    -- SIP/steph-081d9058 is ringing

    -- SIP/steph-081d9058 is making progress passing it to Zap/1-1

    -- SIP/steph-081d9058 answered Zap/1-1

  == Spawn extension (PSTN, s, 3) exited non-zero on 'Zap/1-1'

    -- Hungup 'Zap/1-1'

stksrv02*CLI>

 

 

But if the user drop the call before the SIP/steph answer, my zap channel
seem to lost the connection and I need to remove the cable and replug it
before it can accept incoming call from pstn

 

Any idea why this? Is there a way asterisk can answer the call immediately
rather than after 3 rings

 

Regards

Stephane 

 

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