Geoff, I believe its actually TIMEOUT(absolute)=value. The function name is case sensitive.
- Logan On Dec 30, 2012 9:53 AM, "Geoff Lane" <[email protected]> wrote: > Hi All, > > Asterisk 1.4.22.1 on CentOS 5 > > I've configured my dialplan to limit the maximum call length on > outgoing calls. I've done this as I get the first hour of each call > free with my bundle but I pay through the nose if the call goes over > an hour. > > Family members who live overseas sometimes ask me to transfer them to > UK landline numbers, which is fine by me as it doesn't cost me > provided they don't exceed the hour limit. However, I noticed a few > days ago that a call from my son (who lives in Australia) that I > transferred didn't time out. > > Relevant snippets of extensions.conf follow. > > The incoming (via SIP) call fetches up at the following: > exten => [munged],1,Goto(main,1) > > exten => main,1,Log(NOTICE, Prefilter: call from ${CALLERID(num)}) > exten => main,n,PrivacyManager(2,10) > exten => main,n,GotoIf($["${PRIVACYMGRSTATUS}" = "FAILED"]?withheld,1) > exten => main,n,Log(NOTICE, Incoming call from ${CALLERID(num)}) > exten => main,n,GotoIf($[${BLACKLIST()}]?banned,1) > exten => main,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) > exten => main,n,Dial(${rgMain},${RINGTIME},t) > exten => main,n,Log(NOTICE, Call from ${CALLERID(num)} sent to voicemail) > exten => main,n,VoiceMail(main@default) > > To transfer the call, I press # then dial the number, which is in the > form of 01nnn nnnnnn, and so should fetch up at the following: > exten => _01.,1,SET(Timeout(absolute)=3540) > exten => _01.,n,Dial(${UKGeographical}/${EXTEN},,g) ; send anything > preceded with 01 to UKGeographical > > Am I missing something (e.g. Timeout(absolute) doesn't apply to > transferred calls) or can anyone spot something else that's allowing > the call to continue past the 59 minute set limit? > > TIA, > > -- > Geoff > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
