[asterisk-users] How to use Grandstream GX-2000 phones for paging

Barry D. Hassler Barry.Hassler at hcst.com
Sun Sep 10 10:19:07 MST 2006


Firmware 1.1.1.9? Where's that? most recent on Grandstream's site that I
see is 1.1.0.16, which I have loaded.

On Fri, 2006-09-08 at 20:20 -0500, Lacy Moore - Aspendora wrote:
> 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.
> 
> This works fine for me on firmware 1.1.1.9.
>  
> On 9/8/06, Barry D. Hassler <Barry.Hassler at hcst.com> 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/ 
>           
>         barry.hassler at hcst.com 
>                           +1
>         937-427-9000        
>              +1 937-427-8706
>                      FAX    
>              FWD: 3934279000
>                    (655480) 
>            HCST*Net Support Issues: please email support at hcst.net 
>                 Billing Issues: Please email billing at hcst.net
>         
>         
>         _______________________________________________
>         --Bandwidth and Colocation provided by Easynews.com --
>         
>         asterisk-users mailing list 
>         To UNSUBSCRIBE or update options visit:
>           http://lists.digium.com/mailman/listinfo/asterisk-users
>         
>         
> 
> 
> 
> 
> -- 
> Lacy Moore
> Aspendora, Inc. 


________________________________________________________________________
Barry D. Hassler
President 

HCST
2332 Grange Hall Road
Beavercreek, Ohio
45431-2345
http://www.hcst.net/
 
 barry.hassler at hcst.com 
 +1 937-427-9000        
 +1 937-427-8706 FAX    
         FWD: 3934279000
               (655480) 
        HCST*Net Support Issues: please email support at hcst.net 
             Billing Issues: Please email billing at hcst.net
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060910/bb28e010/attachment.htm


More information about the asterisk-users mailing list