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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven P. Donegan Sent: 12 September 2004 16:55 To: [EMAIL PROTECTED]; 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
