[Asterisk-Users] Grandstream Budgetone 100 Caller ID
shows extension, not incoming Caller ID
Steven P. Donegan
steve at donegan.org
Sun Sep 12 08:54:38 MST 2004
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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