[Asterisk-Users] Grandstream BugetTone 100 Caller IDshows extension, not incoming Caller ID

[David J Carter]

Steven,

On mine in the UK the sip.conf entries are like yours but without the
callerid= entry and my CS phones give me the received callerid fine.


Regards

Dave

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Steven P.
Donegan
Sent: 12 September 2004 16:55
To: eric at fnords.org; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Grandstream Budgetone 100 Caller IDshows
extension, not incoming Caller ID


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