I added the configuration as you suggest but now the phone does not do intercom. I tried Dial and Page in the gxp2000 but everything goes out as Dial.
Here is the extensions.conf now exten=s,1,SIPAddHeader(Alert-Info: <http://127.0.0.1>\;info=Family) exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?3:4) 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(${ARG1}) Any idea? I am very bad on this asterisk thing, sorry guys. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Thursday, August 07, 2008 12:10 PM To: [email protected] Subject: Re: [asterisk-users] intercom/paging with grandstream gxp2000 On 10:59, Thu 07 Aug 08, Fidel Garcia wrote: > Thanks for your reply! > > Just so you have a better understanding of what I am trying to accomplish. > The distinctive ring is working fine with "Family", however, the intercom > configuration that I am currently testing makes all my calls and intercom > call. It does not matter if I call using Dial or Page on the GXP2000, the > call is always and intercom call. For some reason the GXP2000 is receiving > the SipAddHeader when I do Dial and Page. Can you tell what is wrong with > the configuration by looking at the configuration below? > > exten=s,1,SIPAddHeader(Alert-Info: <http://127.0.0.1>\;info=Family) > exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?2:3) > exten=s,3,SIPAddHeader(Call-Info: answer-after=0) if the sip header Call-Info has value answer-after=0 it goes to prio 2, otherwise 3 Now let's have a closer look at those. Hhmm, prio two is the gotoif, prio three adds the answer-after=0 ... I think you mean: exten=s,2,GotoIf($["${SIP_HEADER(Call-Info)}"="answer-after=0"]?3:4) > exten=s,4,Dial(${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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.5.12/1596 - Release Date: 8/6/2008 > 4:55 PM > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer aficionados are both called users?" _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.12/1597 - Release Date: 8/7/2008 5:54 AM _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
