I do not see the L() option on that Dial line. > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of > salaheddine elharit > Sent: Monday, July 11, 2011 4:36 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] timeout with outbound calls > > the CLI show this : > > > -- Executing [0678922645@agents:1] Set("SIP/223-6ec45a88", > "CALLERID(number) > =520460587") in new stack > -- Executing [0678922645@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@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@agents:1] GotoIf("SIP/223-6ec45a88", > "1?3:2") in new stack > -- Goto (agents,h,3) > -- Executing [h@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 <[email protected]> > > > > 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/> <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 <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 > > > >
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