[asterisk-users] intercom/paging with grandstream gxp2000

[Michiel van Baak]

On 10:59, Thu 07 Aug 08, Fidel Garcia wrote:
> 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)

if the sip header Call-Info has value answer-after=0 it goes to prio 2,
otherwise 3

Now let's have a closer look at those.
Hhmm, prio two is the gotoif, prio three adds the answer-after=0 ...

I think you mean:

exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?3:4)

> 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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Michiel van Baak
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