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).
Ok, actually it's under the settings for the Lines and is called: Allow Auto Answer by Call-Info:
Make sure Yes is selected here.
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:
Turn off speaker on remote disconnect: be set to Yes as well.
On 9/8/06, Barry D. Hassler <[EMAIL PROTECTED]> wrote:
This isn't working for me either. I was about to ask this same question, but discovered this recent thread.
I have the following set up in my extensions.conf file, as per Granstream instructions:
[macro-page-grandstream]
exten => s,1,ChanIsAvail(${ARG1}|js); j is for jump, s is for ANY call
exten => s,2,SIPAddHeader(Call-Info: answer-after=0)
exten -> s,3,Dial(${ARG1})
exten => s,4,NoOp();
exten => s,5,Hangup
exten => s,102,NoOp(102) ; Channel not available
exten => s,103,Hangup
[intercoms]
exten => **2311,1,Macro(page-grandstream,SIP/2311)
exten => **2311,2,Hangup
And in my local context:
include => intercoms
When I dial **2311, I see the following debug output:
[Sep 8 15:24:37] -- Starting simple switch on 'Zap/4-1'
[Sep 8 15:24:43] -- Executing SetMusicOnHold("Zap/4-1", "default") in new stack
[Sep 8 15:24:43] -- Executing Goto("Zap/4-1", "intern-hcst-post|**2311|1") in new stack
[Sep 8 15:24:43] -- Goto (intern-hcst-post,**2311,1)
[Sep 8 15:24:43] -- Executing Macro("Zap/4-1", "page-grandstream|SIP/2311") in new stack
[Sep 8 15:24:43] -- Executing ChanIsAvail("Zap/4-1", "SIP/2311|js") in new stack
[Sep 8 15:24:43] -- Executing SIPAddHeader("Zap/4-1", "Call-Info: answer-after=0") in new stack
[Sep 8 15:24:43] -- Executing Hangup("Zap/4-1", "") in new stack
[Sep 8 15:24:43] == Spawn extension (intern-hcst-post, **2311, 2) exited non-zero on 'Zap/4-1'
[Sep 8 15:24:43] -- Hungup 'Zap/4-1'
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.
The BLF functions don't seem to be working either.
I'm running asterisk 1.2.9.1, and have the Granstream GXP2000 reports:
Software Version: Program-- 1.1.0.16 Bootloader-- 1.1.0.1
On Sat, 2006-09-02 at 20:31 -0500, Larry Alkoff wrote:Nic Bellamy wrote: > Zeeshan Zakaria wrote: > >> My client has all Grandstream GX-2000 phones in his office and he >> wants receptionist to use them for paging as well. Currently they are >> using Nortel and receptionist can easily do paging. He said that he >> had somebody setup their old Asterisk system in a way, that >> receptionist could dial an extension, after which her voice was heard >> on all grandstream phones' speaker phones. >> >> I want to know how to setup this type of feature on grandstream >> phones, i.e. dialing an extension will activate all phones' speaker >> phones. > > http://www.grandstream.com/FAQ/Asterisk.htm > > There's a PDF there that tells you (a) what settings to put on the > phone, and (b) how to configure Asterisk to sent the SIP header that > tells the phone to auto-answer. > > Cheers, > Nic. >Please let me know if you get this working. I couldn't. Larry
Barry D. Hassler
President
HCST
2332 Grange Hall Road
Beavercreek, Ohio 45431-2345
http://www.hcst.net/[EMAIL PROTECTED]
+1 937-427-9000
+1 937-427-8706 FAX
FWD: 3934279000 (655480)HCST*Net Support Issues: please email [EMAIL PROTECTED]Billing Issues: Please email [EMAIL PROTECTED]
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