<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 14 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
span.EmailStyle17
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-family:"Calibri","sans-serif";}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</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'>You could also try putting a Progress() statement between Answer and Wait. I know there is a latency issue with DAHDI calls; 5 seconds may or may not be enough for googlevoice.<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>white hat<br><b>Sent:</b> Tuesday, December 06, 2011 3:05 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] google voice calling dial plan question.<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>dwa<br><br>As part of the troubleshooting I updated all of the asterisk packages from the repo with yum. I'm using freepbx distro (centos based) with asterisk 1.8 There were several newer asterisk 1.8 packages available. I'm not using any custom modules in freepbx. After the updates, I restarted asterisk with core restart now but this hasn't helped.<br><br>I'm sure it's a dial plan configuration issue.<br><br>Would you be willing to post sanitized versions of your jabber.conf, gtalk.conf and details regarding the context you're using and how your inbound route is configured in your dial plan?<br><br>Are you using STUN? Is Asterisk behind a NAT device or on a public IP?<br><br>Thanks<o:p></o:p></p><div><p class=MsoNormal>On Tue, Dec 6, 2011 at 10:36 AM, Dave Aibel <<a href="mailto:daibel@pervasivetelecom.com">daibel@pervasivetelecom.com</a>> wrote:<o:p></o:p></p><div><p class=MsoNormal>On Sat, Dec 3, 2011 at 12:59 AM, white hat <<a href="mailto:whitehat238@gmail.com">whitehat238@gmail.com</a>> wrote:<br>> When a caller calls my google voice phone number, I must answer, wait and<br>> press one to accept. Sometimes even that does not work.<br>><br>><o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'>> I just need a little advice on how to write the dial plan. I still have<br>> much to learn about asterisk, and appreciate any advice.<br>><br><br><o:p></o:p></p></div><p class=MsoNormal>Geez,<br><br>Maybe I am just brute forcing it, but, the following dialplan seems to<br>work (at least, most of the time!):<br><br>[gtalk_incoming]<br><br>exten => s,1,Answer()<br>exten => s,n,Wait(5)<br>exten => s,n,SendDTMF(1)<br><br>exten => s,n,Dial(SIP/Ciscofficephone,10)<br>exten => s,n,Playback(vm-nobodyavail)<br>exten => s,n,Playback(vm-pls-try-again)<br>same => n,Hangup()<br><br>HTH,<br><br>dwa<br><br><a href="mailto:daibel@pervasivetelcom.com">daibel@pervasivetelcom.com</a><br><span style='color:#888888'><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></span><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>