asterisk users - Jul 2011 - 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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110708/ac22bc19/attachment.htm>

On Friday 08 Jul 2011, salaheddine elharit wrote:
> 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();
What have you got in your "T" extension?  When the absolute timeout
expires,
it will jump here.

-- 
AJS

Answers come *after* questions.

what can i do in order to fix this issue

regards

2011/7/8 A J Stiles <asterisk_list at earthshod.co.uk>
>  On Friday 08 Jul 2011, salaheddine elharit wrote:
> > 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();
>
> What have you got in your "T" extension?  When the absolute
timeout
> expires,
> it will jump here.
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _____________________________________________________________________
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110708/8813b092/attachment-0001.htm>

On Friday 08 Jul 2011, salaheddine elharit wrote:
> what can i do in order to fix this issue
If and when an absolute timeout occurs, Asterisk jumps to the "T"
extension.
So, in the same context as your 223 extension, you need something like

exten => T,1,NoOp(Absolute timeout triggered)
exten => T,n,Hangup()

This will write "Absolute timeout triggered" to the console and hang
up.  Put
whatever commands in there you like, obviously .....

-- 
AJS

Answers come *after* questions.

> -----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]