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

Steve Maroney steve at stevenet.net
Sun Sep 12 09:24:41 MST 2004


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 :-)
>
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