<br><br><div class="gmail_quote">On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil <span dir="ltr"><<a href="mailto:kurianmthayil@gmail.com">kurianmthayil@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>Hi All,</div>
<div> </div>
<div>Is there a way that I can include call dialing functionality in a webinterface. I have EyeBeam configured with a SIP user say </div>
<div>8440. Will I be able to design an inteface which agent can choose a number and the Dial without punching in the number in </div>
<div>Eyebeam.</div>
<div>I tried using the .call file. ie The agent can choose which number to dial from a web interface. Then, a .call file is </div>
<div>created with the following informations.</div>
<div> </div>
<div>Channel: Zap/g2/9444204943<br>Context: inbound_support<br>Extension: 8440<br>Priority: 0</div>
<div> </div>
<div>Now, in the extensions.conf file, I mentioned the following under inbound_support context.</div>
<div> </div>
<div>[inbound_support]<br>exten =>8440,1,Dial(SIP/8440,55,tTo)<br>exten =>8440,2,Answer<br>exten =>8440,3,Hangup</div>
<div> </div>
<div>But, here the call gets connected only when the receiver end receives the call. When the receiver end picks up the phone, </div>
<div>SIP/8440 rings.</div>
<div> </div>
<div>Is there any other way to implement this. I am not ready to use Vicidial (AstGUIClient) because the interface to be designed </div>
<div>is too custom and the agent should have the list of numbers in front of them while they dial which cannot be done using </div>
<div>Vicicial.</div>
<div> </div>
<div>Regards,</div>
<div> </div>
<div>Kurian Mathew Thayil.</div></blockquote><div><br>The following will ring the internal support personnel (8440) first, after answered, it will then dial the customer (<span style="font-family: monospace;">14109850123</span>) (Are you in Maryland?)<br>
<br>Turn on auto-answer and it should be seamless.<br><span style="font-family: monospace;">
</span><br> <br>Stolen from Wiki:<br><br>To create a call to 14109850123 on a SIP phones called bt101,
here's the file you'd create in /var/spool/asterisk/outgoing (whatever
name is good, of course must be accessible and deletable by asterisk
GNU/Linux user):
<span style="font-family: monospace;"><br><br>Channel: SIP/8440
</span><span style="font-family: monospace;"><br>MaxRetries: 1
</span><span style="font-family: monospace;"> <br>RetryTime: 60
</span><span style="font-family: monospace;"><br>WaitTime: 30
</span><span style="font-family: monospace;"></span></div><div><br><span style="font-family: monospace;">#<br></span><span style="font-family: monospace;"># Assuming that your outgoing call logic is kept in the
</span><span style="font-family: monospace;"># context called [outgoing]
</span><span style="font-family: monospace;"><br>#
</span><span style="font-family: monospace;">Context: outgoing
</span><span style="font-family: monospace;"><br># Extension: 14109850123
</span><span style="font-family: monospace;"><br># Priority: 1
</span></div></div><br clear="all"><br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br><br>