<div dir="ltr"><div>the CLI show this :   </div>
<div> </div>
<div> </div>
<div> -- Executing [0678922645@agents:1] Set(&quot;SIP/223-6ec45a88&quot;, &quot;CALLERID(number)                                                                             =520460587&quot;) in new stack<br>    -- Executing [0678922645@agents:2] MixMonitor(&quot;SIP/223-6ec45a88&quot;, &quot;zap_g1_06                                                                             78922645_1310376223.93960.wav|av(0}V(0)&quot;) in new stack<br>
  == Begin MixMonitor Recording SIP/223-6ec45a88<br>    -- Executing [0678922645@agents:3] Dial(&quot;SIP/223-6ec45a88&quot;, &quot;Zap/g1/06789226                                                                             45|30|A(this-call-may-be-monitored-or-recorded)&quot;) 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 &#39;Zap/1-1&#39;<br>  == Spawn extension (agents, 0678922645, 3) exited non-zero on &#39;SIP/223-6ec45a88&#39;<br>    -- Executing [h@agents:1] GotoIf(&quot;SIP/223-6ec45a88&quot;, &quot;1?3:2&quot;) in new stack<br>
    -- Goto (agents,h,3)<br>    -- Executing [h@agents:3] Hangup(&quot;SIP/223-6ec45a88&quot;, &quot;&quot;) in new stack<br>  == Spawn extension (agents, h, 3) exited non-zero on &#39;SIP/223-6ec45a88&#39;<br>  == End MixMonitor Recording SIP/223-6ec45a88<br>
srvradio*CLI&gt;<br><br><br></div>
<div class="gmail_quote">2011/7/8 Eric Wieling <span dir="ltr">&lt;<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>&gt;</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>&gt; -----Original Message-----<br>&gt; From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>&gt; [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of<br>
&gt; salaheddine elharit<br></div>&gt; Sent: Friday, July 08, 2011 10:23 AM<br>
<div class="im">&gt; To: Asterisk Users Mailing List - Non-Commercial Discussion<br></div>&gt; Subject: Re: [asterisk-users] timeout with outbound calls<br>
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<div></div>
<div class="h5">&gt;<br>&gt; i have tested this solution and i have the same issue<br>&gt;<br>&gt; in my case want to call a phone number 06xxxxxxxx from my<br>&gt; snom phone (sip223)<br>&gt;<br>&gt; the issue still the same<br>
&gt;<br>&gt; any help please<br>&gt;<br>&gt;<br>&gt; 2011/7/8 Eric Wieling &lt;<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>&gt;<br>&gt;<br>&gt;<br>&gt;<br>&gt;<br>&gt;       &gt; -----Original Message-----<br>
&gt;       &gt; From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>&gt;       &gt; [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of<br>
&gt;       &gt; salaheddine elharit<br>&gt;       &gt; Sent: Friday, July 08, 2011 6:43 AM<br>&gt;       &gt; To: Asterisk Users Mailing List - Non-Commercial Discussion<br>&gt;       &gt; Subject: [asterisk-users] timeout with outbound calls<br>
&gt;<br>&gt;       &gt;<br>&gt;       &gt; Hi<br>&gt;       &gt;<br>&gt;       &gt; i want to use timeout  with asterisk 1.4 in order to hangup<br>&gt;       &gt; the outbound calls after 25 sec<br>&gt;       &gt;<br>&gt;       &gt; i call my mobile number 067xxxxxxx from my sip acount 223<br>
&gt;       &gt; and i want to hangu up the call automatic after 25 sec  but<br>&gt;       &gt; there is no hangup after 25<br>&gt;       &gt;<br>&gt;       &gt; could you please help me<br>&gt;       &gt;<br>&gt;       &gt; exten =&gt; 223,1,Set(TIMEOUT(absolute)=25) exten =&gt;<br>
&gt;       &gt; 223,n,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))<br>&gt;       &gt; exten =&gt; 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)<br>&gt;       &gt; exten =&gt; 223,n,Dial(SIP/${EXTEN},,KkTt)<br>&gt;       &gt; exten =&gt; 223,n,Hangup();<br>
&gt;       &gt;<br>&gt;       &gt; Best Regards.<br>&gt;       &gt;<br>&gt;<br>&gt;<br>&gt;       pbx*CLI&gt; core show application dial<br>&gt;<br>&gt;        -= Info about application &#39;Dial&#39; =-<br>&gt;<br>&gt;       [Synopsis]<br>
&gt;       Attempt to connect to another device or endpoint and<br>&gt; bridge the call.<br>&gt;       [snip]<br>&gt;          L(x[:y[:z]]):<br>&gt;              x - Maximum call time, in milliseconds<br>&gt;              y - Warning time, in milliseconds<br>
&gt;              z - Repeat time, in milliseconds<br>&gt;          Limit the call to &lt;x&gt; milliseconds. Play a warning<br>&gt; when &lt;y&gt; mill<br>&gt;          iseconds are left. Repeat the warning every &lt;z&gt;<br>
&gt; milliseconds until time<br>&gt;          expires.<br>&gt;          This option is affected by the following variables:<br>&gt;              ${LIMIT_PLAYAUDIO_CALLER}:<br>&gt;                  yes<br>&gt;                  no<br>
&gt;                  If set, this variable causes Asterisk to play the<br>&gt;                  prompts to the caller.<br>&gt;              ${LIMIT_PLAYAUDIO_CALLEE}:<br>&gt;                  yes<br>&gt;                  no<br>
&gt;                  If set, this variable causes Asterisk to play the<br>&gt;                  prompts to the callee.<br>&gt;              ${LIMIT_TIMEOUT_FILE}:<br>&gt;                  filename<br>&gt;                  If specified, &lt;filename&gt; specifies the sound prompt<br>
&gt;                  to play when the timeout is reached. If not<br>&gt; set, the time remaining<br>&gt;                  will be announced.<br>&gt;              ${LIMIT_CONNECT_FILE}:<br>&gt;                  filename<br>
&gt;                  If specified, &lt;filename&gt; specifies the sound prompt<br>&gt;                  to play when the call begins. If not set,<br>&gt; the time remaining will<br>&gt;                  be announced.<br>
&gt;              ${LIMIT_WARNING_FILE}:<br>&gt;                  filename<br>&gt;                  If specified, &lt;filename&gt; specifies the sound prompt<br>&gt;                  to play as a warning when time &lt;x&gt; is<br>
&gt; reached. If not set, the<br>&gt;                  time remaining will be announced.<br>&gt;       [snip]<br>&gt;<br>&gt;<br>&gt;       --<br>&gt;<br>&gt; _____________________________________________________________________<br>
&gt;       -- Bandwidth and Colocation Provided by<br></div></div>&gt; <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> &lt;<a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com/</a>&gt;  --<br>

<div>
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