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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 1.2.9.1, and have the Granstream GXP2000 reports:<BR>
<B>Software Version: </B> Program-- 1.1.0.16 Bootloader-- 1.1.0.1<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 HREF="http://www.grandstream.com/FAQ/Asterisk.htm">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>
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