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

Lacy Moore - Aspendora aspendora at gmail.com
Fri Sep 8 18:20:59 MST 2006


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
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>
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-- 
Lacy Moore
Aspendora, Inc.
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