Show us the CLI output of the failed call. > -----Original Message----- > From: [email protected] > [mailto:[email protected]] 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 <[email protected]> > > > > > > -----Original Message----- > > From: [email protected] > > [mailto:[email protected]] 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/> -- > 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 > > > >
-- _____________________________________________________________________ -- 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
