<div>You also have to make sure that on the web config for Grandstream that you allow it to receive auto-answer (or something to that effect).</div>
<div> </div>
<div>Ok, actually it's under the settings for the Lines and is called: <em>Allow Auto Answer by Call-Info: </em></div>
<div><em></em> </div>
<div>Make sure Yes is selected here.</div>
<div> </div>
<div>You can use what Barry has below for paging (or rather intercom) to a single phone. For actual paging (i.e., several phones), use the Page command (show application page for options from the CLI). On paging, I would recommend this:
<em>Turn off speaker on remote disconnect: </em>be set to Yes as well.</div>
<div><br>This works fine for me on firmware <a href="http://1.1.1.9">1.1.1.9</a>.<br> </div>
<div><span class="gmail_quote">On 9/8/06, <b class="gmail_sendername">Barry D. Hassler</b> <<a href="mailto:Barry.Hassler@hcst.com">Barry.Hassler@hcst.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div>
<div>This isn't working for me either. I was about to ask this same question, but discovered this recent thread.<br><br>I have the following set up in my extensions.conf file, as per Granstream instructions:<br><tt><b>[macro-page-grandstream]
</b></tt><br><tt><b>exten => s,1,ChanIsAvail(${ARG1}|js); j is for jump, s is for ANY call</b></tt><br><tt><b>exten => s,2,SIPAddHeader(Call-Info: answer-after=0)</b></tt><br><tt><b>exten -> s,3,Dial(${ARG1})</b>
</tt><br><tt><b>exten => s,4,NoOp();</b></tt><br><tt><b>exten => s,5,Hangup</b></tt><br><tt><b>exten => s,102,NoOp(102) ; Channel not available</b></tt><br><tt><b>exten => s,103,Hangup</b></tt><br><br>[intercoms]
<br><b><tt>exten => **2311,1,Macro(page-grandstream,SIP/2311)</tt></b><br><b><tt>exten => **2311,2,Hangup</tt></b><br><br>And in my local context:<br><b><tt>include => intercoms</tt></b><br><br>When I dial **2311, I see the following debug output:
<br><tt><b>[Sep 8 15:24:37] -- Starting simple switch on 'Zap/4-1'</b></tt><br><tt><b>[Sep 8 15:24:43] -- Executing SetMusicOnHold("Zap/4-1", "default") in new stack</b></tt><br><tt><b>[Sep 8 15:24:43] -- Executing Goto("Zap/4-1", "intern-hcst-post|**2311|1") in new stack
</b></tt><br><tt><b>[Sep 8 15:24:43] -- Goto (intern-hcst-post,**2311,1)</b></tt><br><tt><b>[Sep 8 15:24:43] -- Executing Macro("Zap/4-1", "page-grandstream|SIP/2311") in new stack</b></tt><br>
<tt><b>[Sep 8 15:24:43] -- Executing ChanIsAvail("Zap/4-1", "SIP/2311|js") in new stack</b></tt><br><tt><b>[Sep 8 15:24:43] -- Executing SIPAddHeader("Zap/4-1", "Call-Info: answer-after=0") in new stack
</b></tt><br><tt><b>[Sep 8 15:24:43] -- Executing Hangup("Zap/4-1", "") in new stack</b></tt><br><tt><b>[Sep 8 15:24:43] == Spawn extension (intern-hcst-post, **2311, 2) exited non-zero on 'Zap/4-1'
</b></tt><br><tt><b>[Sep 8 15:24:43] -- Hungup 'Zap/4-1'</b></tt><br><br>Is this a problem with the SIPAddHEader that it is jumping immediately to Hangup? I see NO SIP traffic as a result of this, and sip debug shows nothing out of the ordinary.
<br><br>The BLF functions don't seem to be working either.<br><br>I'm running asterisk <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://1.2.9.1/" target="_blank">1.2.9.1</a>, and have the Granstream GXP2000 reports:
<br><b>Software Version: </b> Program-- <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://1.1.0.16/" target="_blank">1.1.0.16</a> Bootloader-- <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://1.1.0.1/" target="_blank">
1.1.0.1</a></div>
<div><span class="q"><br><br><br>On Sat, 2006-09-02 at 20:31 -0500, Larry Alkoff wrote:
<blockquote type="CITE"><pre><font color="#000000">Nic Bellamy wrote:</font>
<font color="#000000">> Zeeshan Zakaria wrote:</font>
<font color="#000000">> </font>
<font color="#000000">>> My client has all Grandstream GX-2000 phones in his office and he </font>
<font color="#000000">>> wants receptionist to use them for paging as well. Currently they are </font>
<font color="#000000">>> using Nortel and receptionist can easily do paging. He said that he </font>
<font color="#000000">>> had somebody setup their old Asterisk system in a way, that </font>
<font color="#000000">>> receptionist could dial an extension, after which her voice was heard </font>
<font color="#000000">>> on all grandstream phones' speaker phones.</font>
<font color="#000000">>> </font>
<font color="#000000">>> I want to know how to setup this type of feature on grandstream </font>
<font color="#000000">>> phones, i.e. dialing an extension will activate all phones' speaker </font>
<font color="#000000">>> phones.</font>
<font color="#000000">> </font>
<font color="#000000">> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.grandstream.com/FAQ/Asterisk.htm" target="_blank">http://www.grandstream.com/FAQ/Asterisk.htm</a></font>
<font color="#000000">> </font>
<font color="#000000">> There's a PDF there that tells you (a) what settings to put on the </font>
<font color="#000000">> phone, and (b) how to configure Asterisk to sent the SIP header that </font>
<font color="#000000">> tells the phone to auto-answer.</font>
<font color="#000000">> </font>
<font color="#000000">> Cheers,</font>
<font color="#000000">> Nic.</font>
<font color="#000000">> </font>
<font color="#000000">Please let me know if you get this working. I couldn't.</font>
<font color="#000000">Larry</font>
</pre></blockquote></span></div>
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