[asterisk-users] intercom/paging with grandstream gxp2000

[Fidel Garcia]

Thanks for your reply!

Just so you have a better understanding of what I am trying to accomplish.
The distinctive ring is working fine with "Family", however, the intercom
configuration that I am currently testing makes all my calls and intercom
call. It does not matter if I call using Dial or Page on the GXP2000, the
call is always and intercom call. For some reason the GXP2000 is receiving
the SipAddHeader when I do Dial and Page. Can you tell what is wrong with
the configuration by looking at the configuration below?

exten=s,1,SIPAddHeader(Alert-Info: <http://127.0.0.1>\;info=Family)
exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?2:3)
exten=s,3,SIPAddHeader(Call-Info: answer-after=0)
exten=s,4,Dial(${ARG2},20)
exten=s,5,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})

what would you do differently?



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gordon
Henderson
Sent: Thursday, August 07, 2008 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000

On Wed, 6 Aug 2008, Fidel Garcia wrote:

> Guys I have been reading for days on how to get this to work with asterisk
> and for some reason every time I call the call goes to intercom.  I know I
> must be doing something wrong with the way I am adding the steps to my
call;
> I am not familiar with variables and flags.

What *exactly* are you trying to achieve?

I have used both paging and intercom mode in the Grandstreams with good 
results.

You do need the settings in the phone set ON - ie.

 	Allow Auto Answer by Call-Info:   No      Yes
 	Turn off speaker on remote disconnect:   No      Yes

These both need to be set to YES or ON.

That won't affect normal calls to that account on the phone - although the 
"turn off speaker" one does make the phone easier to use IMO...

So call the phone and the person answers normally, as before, but if you 
rhen add the SIP header:

 	SIPAddHeader(Call-Info: answer-after=0)

The phone will auto-answer - when the next Dial or Page command is 
directed to it.

What next? If you want to Page the phone, use the Page() application.

So if the phone is SIP/100 then to Dial the phone normally..

     exten => 100,1,Dial(SIP/100)

but to page it:

     exten => 200,1,SIPAddHeader(Call-Info: answer-after=0)
     exten => 200,n,Page(SIP/100)

and to intercom to it:

     exten => 300,1,SIPAddHeader(Call-Info: answer-after=0)
     exten => 300,n,Page(SIP/100,d)


So this has added 3 new extensions, 100, 200 and 300 - which all 'call' 
SIP/100, but in 3 differet ways.

Gordon

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