[Asterisk-Users] Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID

[Steve Maroney]

Im not sure if you mentioned this or not, but what is your call
originatiing from ? On an FXO port ? Make sure your zapata.conf file is
correct. There are a lot of callerid options in there.

Thank you,
Steve Maroney

On Sun, 12 Sep 2004, Steven P. Donegan wrote:

> Firmware now current (1.0.5.11) - no change in what is displayed on the
> phone. Good thought though :-)
>
> Steve Maroney wrote:
>
> >Try upgrading the firmware
> >
> >Thank you,
> >Steve Maroney
> >
> >On Sun, 12 Sep 2004, Steven P. Donegan wrote:
> >
> >
> >
> >>Eric Wieling wrote:
> >>
> >>
> >>
> >>>On Sun, 2004-09-12 at 09:41, Duane wrote:
> >>>
> >>>
> >>>
> >>>
> >>>>Steven P. Donegan wrote:
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>>I've looked through the archives - and see questions similar to mine,
> >>>>>but no answers. What, if anything, can be done to get the incoming
> >>>>>Caller ID to be presented on the Budgetone's Caller ID display? In all
> >>>>>other respects the phone+Asterisk seem to be extremely happy with each
> >>>>>other.
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>What you need to do is strip the alpha caller name from the caller ID,
> >>>>the 101's can only handle numbers and it's trying to display a name...
> >>>>
> >>>>
> >>>>
> >>>>
> >>>I don't think this is the problem. If it was a general problem hundreds
> >>>f people would be complaining about this. Put a
> >>>NoOp(CALLERID=${CALLERID}) in the dialplan just before the Dial line to
> >>>ring the GS phone.  What you should see is something like CALLERID=Bob
> >>>Dobbs <666> on the console when the NoOp runs.  If you see ANYTHING that
> >>>isn't in the format of Caller*ID Name <calleridnumber. then you have
> >>>something messed up in your Asterisk config.  As said, the BT101 only
> >>>can display Caller*ID numbers, it should generally just throw out the
> >>>Caller*ID name.  You don't mention what COUNTRY you are in so I don't
> >>>know if it's an issue between what your telco sends and what Asterisk
> >>>expects.  In the USA this is not an issue, in other countries it *could*
> >>>be an issue.
> >>>
> >>>
> >>>
> >>>
> >>>
> >>I am in the US, and caller ID otherwise works fine (ie on analog
> >>stations it comes thorough just fine).
> >>
> >>sip.conf configlet:
> >>
> >>[1000]
> >>type=friend
> >>username=1000
> >>fromuser=1000
> >>callerid=Computer Room <1000>
> >>host=dynamic
> >>nat=no
> >>canreinvite=yes
> >>dtmfmode=info
> >>mailbox=1000 at default
> >>disallow=all
> >>allow=ulaw
> >>
> >>extensions.conf configlet:
> >>
> >>[sip-access]
> >>
> >>exten => 1000,1,Macro(stdexten,1000,SIP/1000)
> >>
> >>The stdexten Macro is the vanilla one from 'stock' Asterisk.
> >>
> >>On the console I see all the appropriate caller ID/connection info, and
> >>the Voicemail application definitely emails me the correct stuff - so it
> >>seems it is something being lost between Asterisk/Grandstream...
> >>
> >>Thanks for any help - this is on my home PBX - but once it all works I
> >>will be rolling it out as a test at a friendly beta customer :-)
> >>
> >>_______________________________________________
> >>Asterisk-Users mailing list
> >>Asterisk-Users at lists.digium.com
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >>
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>