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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clive.chan(Alpha Trilogies Networks)
2007-Jul-02 01:16 UTC
[asterisk-users] Asterisk strange behaviour
Stephane,
What does it means?
" 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"
Does it mean that the hangup call will not hangup even the caller has put
hang up the line?
Please provide your zaptel.conf, and zapata.conf file here, so everyone here
can help.
Thank you
Clive
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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