Yes, it worked :D<div><br></div><div>Thankyou guys for the help. <br><br><div class="gmail_quote">2013/3/8 Luis H. Forchesatto <span dir="ltr"><<a href="mailto:luisforchesatto@gmail.com" target="_blank">luisforchesatto@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I think I found the problem. Better looking the sip_additional.conf file I noticed that a few extensions didnt had a callgroup and pickgroup configured, even with the interface appointing otherwise. <div>
<br></div><div>I manually configured this options and reloader asterisk and now I'm gonna test the extensions and see if it works now. </div>
<div><br></div><div>I'll be back with the result soon. </div><div><br></div><div><div><div class="h5"><br><br><div class="gmail_quote">2013/3/8 A J Stiles <span dir="ltr"><<a href="mailto:asterisk_list@earthshod.co.uk" target="_blank">asterisk_list@earthshod.co.uk</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>On Thursday 07 March 2013, Luis H. Forchesatto wrote:<br>
> Greetings.<br>
><br>
> I got an extension on my Elastix who cannot pick calls on the other<br>
> extensions, but It can transfer his calls to the other extensions. When<br>
> this extension tries to pickup a call pressing *8 it simply does not pick<br>
> it up. Transfering calls works just fine so dtmf may be not the problem.<br>
><br>
> Where should I look?<br>
<br>
</div>/etc/asterisk/sip.conf (if it's s SIP phone); otherwise the corresponding<br>
configuration file for whatever technology it is using. Make sure that the<br>
"pickupgroup" for that extension is the same as the other extensions. Then<br>
$ sudo asterisk -x 'reload' (or enter "reload" in Asterisk CLI) to apply the<br>
change.<br>
<span><font color="#888888"><br>
--<br>
AJS<br>
<br>
Answers come *after* questions.<br>
</font></span><div><div><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><br>
</div></div></blockquote></div><br><br clear="all"><div><br></div></div></div><div class="im">-- <br>Att.<b><br></b><b></b><div>Luis H. Forchesatto</div><div>Mail: <a href="mailto:luis_forchesatto@hotmail.com" target="_blank">luis_forchesatto@hotmail.com</a></div>
</div></div>
</blockquote></div><br><br clear="all"><div><br></div>-- <br>Att.<b><br></b><b></b><div>Luis H. Forchesatto</div><div>Mail: <a href="mailto:luis_forchesatto@hotmail.com" target="_blank">luis_forchesatto@hotmail.com</a></div>
</div>