[asterisk-users] intercom/paging with grandstream gxp2000

Gordon Henderson gordon+asterisk at drogon.net
Thu Aug 7 11:07:43 CDT 2008


On Thu, 7 Aug 2008, 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)
> 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?

Well, I'd stop using numbers and start using labels to begin with.

Line 2:

I have no idea what this is doing, but it looks like You're saying that if 
a SIP_HEADER called Call-Info is already set to answer-after=0 then jump 
to ... step 2. This would cause an infinite loop.

then in the next step you explicitly set the Call-Info to answer-after=0, 
so what do you expect?

Your dialplan is just broken.

Create TWO extensions for this phone. One to make the phone ring as a 
normal phone and the other to make the phone go into Intercom mode. You 
can not select modes like this.

I'm assuming that's a macro, if-so, them make it:

[Macro-ringPhone]
exten => s,1,Dial(${ARG2},20)
exten => s,n,Goto(${DIALSTATUS})

exten => s,n(NOANSWER),Voicemail(${ARG1},u)
...and so on...

Then to call it:

exten => 100,1,Macro(ringPhone,100,100)

or to call it in intercom mode.

exten => 200,1,SIPAddHeader(Call-Info: answer-after=0)
exten => 200,n,Macro(ringPhone,100,100)

Gordon


>
>
>
> -----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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