[Asterisk-Users] LiveVoIP Problems?
Jay Milk
jay at skimmilk.net
Mon Mar 7 16:26:29 MST 2005
Afaik, caller-id name is not passed on between lecs and clecs (via SS7)
-- that's what I remember from a thread I read here in the past. That's
why most clecs maintain their own DBs, and that's also why it takes
weeks for callerid information to propagate, and why your name may show
up in different variations (or not at all) depending where you call.
To clarify -- "PSTN termination" to me means "any VOIP service that
terminates to PSTN". I have yet to see an IAX or SIP service provider
that passes on caller-name via PRI or SS7 to the PSTN. And even if they
did, chances of the name arriving at the called number are slim to none.
> -----Original Message-----
> From: Paul Fielding [mailto:paul at fielding.ca]
> Sent: Monday, March 07, 2005 4:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] LiveVoIP Problems?
>
>
> ----- Original Message -----
> From: "Jay Milk" <jay at skimmilk.net>
>
> > You won't be able to send caller-id NAME with any PSTN termination.
> > That's just not how that works. Each CLEC looks up the
> name in some
> > mystical database based on the phone number. How to get that DB, I
> > don't know, but it sure would be nice to integrate
> something like this
> > into *, wouldn't it?
>
> Sure, but wouldn't LiveVoip be using PRI as opposed to PSTN?
> I dunno about
> in the US, but here (Canada) we've got switch-based CallerID
> and user-based
> CallerID. As long as you're using a PRI based line the user
> can fire both
> caller number and caller name to the telco and see the
> results on the other
> side (user-based). If this isn't available in the US then
> that's fine, but
> otherwise it'd be nice to be able to send caller name along
> with the number
> through LiveVoip.
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