asterisk users - Aug 2008 - intercom/paging with grandstream gxp2000

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.

 

Here is my configuration:
Digium Asterisk AA50 with Granstream GXP2000 using the latest firmware.

 

Extensions.conf:

 

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

exten=s,3,Dial(${ARG2},20)

exten=s,4,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})

 

GXP2000 configuration:

Under Account1 I checked options:

 

Allow Auto Answer by Call-Info:   No      Yes  

 

Turn off speaker on 
remote disconnect:   No      Yes

 

 

 

Fidel Garcia

System Engineer

 

sysTeam.

7205 NW 19th Street, Suite 302
Miami, Florida 33126

Email: fgarcia at systeamusa.com 

Tel: (305)-477-7303 Fax: (305)-477-0013 

http://www.systeamusa.com

 

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I am sorry, this is the actual extensions.conf:

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})

 

As you can see here Goto and SIPAddHeader are 2 and 3. In the  prior email I
had both lines under 2.

 

Fidel Garcia

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Fidel Garcia
Sent: Wednesday, August 06, 2008 5:05 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] intercom/paging with grandstream gxp2000

 

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.

 

Here is my configuration:
Digium Asterisk AA50 with Granstream GXP2000 using the latest firmware.

 

Extensions.conf:

 

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

exten=s,3,Dial(${ARG2},20)

exten=s,4,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})

 

GXP2000 configuration:

Under Account1 I checked options:

 

Allow Auto Answer by Call-Info:   No      Yes  

 

Turn off speaker on 
remote disconnect:   No      Yes

 

 

 

Fidel Garcia

System Engineer

 

sysTeam.

7205 NW 19th Street, Suite 302
Miami, Florida 33126

Email: fgarcia at systeamusa.com 

Tel: (305)-477-7303 Fax: (305)-477-0013 

http://www.systeamusa.com

 

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.5.12/1595 - Release Date: 8/6/2008
8:23 AM

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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