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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>It can have to do with either the telephones dial plan or the context in the Asterisk dial plan combined with your features.conf settings.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Ramiro Paz<br><b>Sent:</b> Monday, November 07, 2011 8:46 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Problem with Atxfer for the calling party<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>Hi:<br><br>Same problem here (asterisk doesn't wait all the digits be typed when making a call transfer). Does anybody knows something about this? Thanks in advance.<br><br>Greetings,<br><br clear=all><b><span style='color:#330033'>Ramiro PAZ<br>MASTERLINE LOGISTICS<br><br></span></b><o:p></o:p></p><div><p class=MsoNormal>2011/11/1 Antonio Modesto <<a href="mailto:modesto@isimples.com.br">modesto@isimples.com.br</a>><o:p></o:p></p><div><p class=MsoNormal>Good morning,<br><br> I have not solved this problem yet, but, I found that the source of the problem are my macros. For example, I have this context:<br><br>context ramais {<br>101 => &dial_sip(exten1);<br>102 => &dial_sip(exten2);<br>103 => &dial_sip(exten3);<br>};<br><br>All these extensions use the dial_sip macro, I have changed this context to use the Dial application instead of dial_sip macro, it worked fine. The problem is that when i use the macro, the current context is changed to the dial_sip context, the dial_sip context is automatically created by asterisk when i use any macro and of fact this context doesn't have the ramais context included. Is there some way to specify on which context the macro will run?<o:p></o:p></p><div><div><p class=MsoNormal><br><br>On Mon, 2011-10-31 at 09:09 -0200, Antonio Modesto wrote:<o:p></o:p></p></div></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p class=MsoNormal>Good Morning,<br><br> I have an asterisk18-1.8.7.1 running on a FreeBSD 8.2-STABLE, and it is working well so far, i'm just having some problems with atxfer.<br><br> I have written this macro to dial sip extensions:<br><br>macro dial_sip(exten) {<br> Verbose(2,"==> Chamando a MACRO dial_sip - ponto 1 macros.ael <==");<br> Verbose(4,"====> Macro dial_sip iniciada.");<br> ChanIsAvail(SIP/${exten});<br> Verbose(2,"==> ${AVAILORIGCHAN}");<br><br> if ("${AVAILORIGCHAN}" != "")<br> {<br> Verbose(4,"====> SIP/${exten} parece estar disponivel, vou disca-lo agora.");<br> Set(FromExt=${CALLERID(num)});<br> System(/bin/sh /var/spool/asterisk/calllog/log.sh SIP/${FromExt} SIP/${exten} SIP-TO-SIP);<br> Verbose(4,"====> System status: ${SYSTEMSTATUS}");<br> Dial(SIP/${exten},${SIP_DIAL_TIMEOUT},Ttr);<br> Hangup();<br> }<br> else<br> {<br> Verbose(2,"====> SIP/${exten} nao esta disponivel.");<br> Hangup();<br> };<br><br><br> NoOp("From ${MACRO_EXTEN} to ${exten});<br> System(${CALLLOGDIR}/log.sh ${exten});<br><br> return;<br>};<br><br>It is working, but the calling party is not able to transfer the calls because asterisk doesn't wait all the digits be typed, it tries to transfer the call when the first digit is pressed (We use 3 digits extensions):<br><br>[Oct 31 09:04:01] WARNING[2926]: features.c:2315 builtin_atxfer: Extension '1' does not exist in context 'dial_sip'<br> == Spawn extension (dial_sip, ~~s~~, 11) exited non-zero on 'SIP/modesto-0000000d'<br>[Oct 31 09:04:03] WARNING[2926]: features.c:2319 builtin_atxfer: No digits dialed for atxfer.<br><br>Does anyone have suggestions?<br><br>Regards. <o:p></o:p></p></div></div><pre><span style='color:#888888'>--<o:p></o:p></span></pre><pre><span style='color:#888888'>_____________________________________________________________________<o:p></o:p></span></pre><pre><span style='color:#888888'>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<o:p></o:p></span></pre><pre><span style='color:#888888'>New to Asterisk? Join us for a live introductory webinar every Thurs:<o:p></o:p></span></pre><pre><span style='color:#888888'> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><o:p></o:p></span></pre><pre><span style='color:#888888'><o:p> </o:p></span></pre><pre><span style='color:#888888'>asterisk-users mailing list<o:p></o:p></span></pre><pre><span style='color:#888888'>To UNSUBSCRIBE or update options visit:<o:p></o:p></span></pre><pre><span style='color:#888888'> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></span></pre></blockquote><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>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><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>