[asterisk-users] timeout with outbound calls
salaheddine elharit
salah.elharit200 at gmail.com
Mon Jul 11 03:36:23 CDT 2011
the CLI show this :
-- Executing [0678922645 at agents:1] Set("SIP/223-6ec45a88",
"CALLERID(number)
=520460587") in new stack
-- Executing [0678922645 at agents:2] MixMonitor("SIP/223-6ec45a88",
"zap_g1_06
78922645_1310376223.93960.wav|av(0}V(0)") in new stack
== Begin MixMonitor Recording SIP/223-6ec45a88
-- Executing [0678922645 at agents:3] Dial("SIP/223-6ec45a88",
"Zap/g1/06789226
45|30|A(this-call-may-be-monitored-or-recorded)") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0678922645
-- Zap/1-1 is proceeding passing it to SIP/223-6ec45a88
-- Zap/1-1 is ringing
[Jul 11 09:23:50] NOTICE[30408]: chan_sip.c:15012 handle_request_subscribe:
Rece
ived SIP subscribe for peer without mailbox: 212
-- Zap/1-1 answered SIP/223-6ec45a88
[Jul 11 09:23:51] WARNING[10599]: file.c:607 ast_openstream_full: File
this-call
-may-be-monitored-or-recorded does not exist in any format
[Jul 11 09:23:51] WARNING[10599]: file.c:906 ast_streamfile: Unable to open
this
-call-may-be-monitored-or-recorded (format 0x48 (alaw|slin)): No such file
or
di
rectory
-- Hungup 'Zap/1-1'
== Spawn extension (agents, 0678922645, 3) exited non-zero on
'SIP/223-6ec45a88'
-- Executing [h at agents:1] GotoIf("SIP/223-6ec45a88", "1?3:2") in new
stack
-- Goto (agents,h,3)
-- Executing [h at agents:3] Hangup("SIP/223-6ec45a88", "") in new stack
== Spawn extension (agents, h, 3) exited non-zero on 'SIP/223-6ec45a88'
== End MixMonitor Recording SIP/223-6ec45a88
srvradio*CLI>
2011/7/8 Eric Wieling <EWieling at nyigc.com>
>
> Show us the CLI output of the failed call.
>
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> > salaheddine elharit
> > Sent: Friday, July 08, 2011 10:23 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] timeout with outbound calls
> >
> > i have tested this solution and i have the same issue
> >
> > in my case want to call a phone number 06xxxxxxxx from my
> > snom phone (sip223)
> >
> > the issue still the same
> >
> > any help please
> >
> >
> > 2011/7/8 Eric Wieling <EWieling at nyigc.com>
> >
> >
> >
> >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com
> > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> > > salaheddine elharit
> > > Sent: Friday, July 08, 2011 6:43 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [asterisk-users] timeout with outbound calls
> >
> > >
> > > Hi
> > >
> > > i want to use timeout with asterisk 1.4 in order to hangup
> > > the outbound calls after 25 sec
> > >
> > > i call my mobile number 067xxxxxxx from my sip acount 223
> > > and i want to hangu up the call automatic after 25 sec but
> > > there is no hangup after 25
> > >
> > > could you please help me
> > >
> > > exten => 223,1,Set(TIMEOUT(absolute)=25) exten =>
> > > 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> > > exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> > > exten => 223,n,Dial(SIP/${EXTEN},,KkTt)
> > > exten => 223,n,Hangup();
> > >
> > > Best Regards.
> > >
> >
> >
> > pbx*CLI> core show application dial
> >
> > -= Info about application 'Dial' =-
> >
> > [Synopsis]
> > Attempt to connect to another device or endpoint and
> > bridge the call.
> > [snip]
> > L(x[:y[:z]]):
> > x - Maximum call time, in milliseconds
> > y - Warning time, in milliseconds
> > z - Repeat time, in milliseconds
> > Limit the call to <x> milliseconds. Play a warning
> > when <y> mill
> > iseconds are left. Repeat the warning every <z>
> > milliseconds until time
> > expires.
> > This option is affected by the following variables:
> > ${LIMIT_PLAYAUDIO_CALLER}:
> > yes
> > no
> > If set, this variable causes Asterisk to play the
> > prompts to the caller.
> > ${LIMIT_PLAYAUDIO_CALLEE}:
> > yes
> > no
> > If set, this variable causes Asterisk to play the
> > prompts to the callee.
> > ${LIMIT_TIMEOUT_FILE}:
> > filename
> > If specified, <filename> specifies the sound prompt
> > to play when the timeout is reached. If not
> > set, the time remaining
> > will be announced.
> > ${LIMIT_CONNECT_FILE}:
> > filename
> > If specified, <filename> specifies the sound prompt
> > to play when the call begins. If not set,
> > the time remaining will
> > be announced.
> > ${LIMIT_WARNING_FILE}:
> > filename
> > If specified, <filename> specifies the sound prompt
> > to play as a warning when time <x> is
> > reached. If not set, the
> > time remaining will be announced.
> > [snip]
> >
> >
> > --
> >
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com <http://www.api-digital.com/> --
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> > webinar every Thurs:
> > http://www.asterisk.org/hello
> >
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> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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