<div dir="ltr"><div>the CLI show this :   </div>
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<div> -- Executing [0678922645@agents:1] Set("SIP/223-6ec45a88", "CALLERID(number)                                                                             =520460587") in new stack<br>    -- Executing [0678922645@agents:2] MixMonitor("SIP/223-6ec45a88", "zap_g1_06                                                                             78922645_1310376223.93960.wav|av(0}V(0)") in new stack<br>
  == Begin MixMonitor Recording SIP/223-6ec45a88<br>    -- Executing [0678922645@agents:3] Dial("SIP/223-6ec45a88", "Zap/g1/06789226                                                                             45|30|A(this-call-may-be-monitored-or-recorded)") in new stack<br>
    -- Requested transfer capability: 0x00 - SPEECH<br>    -- Called g1/0678922645<br>    -- Zap/1-1 is proceeding passing it to SIP/223-6ec45a88<br>    -- Zap/1-1 is ringing<br>[Jul 11 09:23:50] NOTICE[30408]: chan_sip.c:15012 handle_request_subscribe: Rece                                                                             ived SIP subscribe for peer without mailbox: 212<br>
    -- Zap/1-1 answered SIP/223-6ec45a88<br>[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<br>
[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<br>
    -- Hungup 'Zap/1-1'<br>  == Spawn extension (agents, 0678922645, 3) exited non-zero on 'SIP/223-6ec45a88'<br>    -- Executing [h@agents:1] GotoIf("SIP/223-6ec45a88", "1?3:2") in new stack<br>
    -- Goto (agents,h,3)<br>    -- Executing [h@agents:3] Hangup("SIP/223-6ec45a88", "") in new stack<br>  == Spawn extension (agents, h, 3) exited non-zero on 'SIP/223-6ec45a88'<br>  == End MixMonitor Recording SIP/223-6ec45a88<br>
srvradio*CLI><br><br><br></div>
<div class="gmail_quote">2011/7/8 Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>></span><br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote"><br>Show us the CLI output of the failed call.<br>
<div class="im"><br>> -----Original Message-----<br>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of<br>
> salaheddine elharit<br></div>> Sent: Friday, July 08, 2011 10:23 AM<br>
<div class="im">> To: Asterisk Users Mailing List - Non-Commercial Discussion<br></div>> Subject: Re: [asterisk-users] timeout with outbound calls<br>
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<div class="h5">><br>> i have tested this solution and i have the same issue<br>><br>> in my case want to call a phone number 06xxxxxxxx from my<br>> snom phone (sip223)<br>><br>> the issue still the same<br>
><br>> any help please<br>><br>><br>> 2011/7/8 Eric Wieling <<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>><br>><br>><br>><br>><br>>       > -----Original Message-----<br>
>       > From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>>       > [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of<br>
>       > salaheddine elharit<br>>       > Sent: Friday, July 08, 2011 6:43 AM<br>>       > To: Asterisk Users Mailing List - Non-Commercial Discussion<br>>       > Subject: [asterisk-users] timeout with outbound calls<br>
><br>>       ><br>>       > Hi<br>>       ><br>>       > i want to use timeout  with asterisk 1.4 in order to hangup<br>>       > the outbound calls after 25 sec<br>>       ><br>>       > i call my mobile number 067xxxxxxx from my sip acount 223<br>
>       > and i want to hangu up the call automatic after 25 sec  but<br>>       > there is no hangup after 25<br>>       ><br>>       > could you please help me<br>>       ><br>>       > exten => 223,1,Set(TIMEOUT(absolute)=25) exten =><br>
>       > 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))<br>>       > exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)<br>>       > exten => 223,n,Dial(SIP/${EXTEN},,KkTt)<br>>       > exten => 223,n,Hangup();<br>
>       ><br>>       > Best Regards.<br>>       ><br>><br>><br>>       pbx*CLI> core show application dial<br>><br>>        -= Info about application 'Dial' =-<br>><br>>       [Synopsis]<br>
>       Attempt to connect to another device or endpoint and<br>> bridge the call.<br>>       [snip]<br>>          L(x[:y[:z]]):<br>>              x - Maximum call time, in milliseconds<br>>              y - Warning time, in milliseconds<br>
>              z - Repeat time, in milliseconds<br>>          Limit the call to <x> milliseconds. Play a warning<br>> when <y> mill<br>>          iseconds are left. Repeat the warning every <z><br>
> milliseconds until time<br>>          expires.<br>>          This option is affected by the following variables:<br>>              ${LIMIT_PLAYAUDIO_CALLER}:<br>>                  yes<br>>                  no<br>
>                  If set, this variable causes Asterisk to play the<br>>                  prompts to the caller.<br>>              ${LIMIT_PLAYAUDIO_CALLEE}:<br>>                  yes<br>>                  no<br>
>                  If set, this variable causes Asterisk to play the<br>>                  prompts to the callee.<br>>              ${LIMIT_TIMEOUT_FILE}:<br>>                  filename<br>>                  If specified, <filename> specifies the sound prompt<br>
>                  to play when the timeout is reached. If not<br>> set, the time remaining<br>>                  will be announced.<br>>              ${LIMIT_CONNECT_FILE}:<br>>                  filename<br>
>                  If specified, <filename> specifies the sound prompt<br>>                  to play when the call begins. If not set,<br>> the time remaining will<br>>                  be announced.<br>
>              ${LIMIT_WARNING_FILE}:<br>>                  filename<br>>                  If specified, <filename> specifies the sound prompt<br>>                  to play as a warning when time <x> is<br>
> reached. If not set, the<br>>                  time remaining will be announced.<br>>       [snip]<br>><br>><br>>       --<br>><br>> _____________________________________________________________________<br>
>       -- Bandwidth and Colocation Provided by<br></div></div>> <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> <<a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com/</a>>  --<br>
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New to Asterisk? Join us for a live introductory webinar every Thurs:<br>              <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
  <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></div></div></blockquote></div><br></div>