[asterisk-users] PRI Outbound CallerID Question

[Kristian Kielhofner]

Shawn Kelley wrote:
> Hi all,
> I've searched around and haven't found much of an answer to my issue. Any
> advice from you would be appreciated.
> 
> Problem: Need to take an inbound call from our PRI and forward it to another
> PSTN user via the PRI, sending the original callers id with it.
> I know this can be done since we currently use an 800 service that does it.
> You call the 800 number; they answer and put you on hold. They then outcall
> to the pstn numbers we have defined and the incoming call shows up with the
> original callers CID, we answer and have options to accept or reject the
> call.
> 
> So I know the 800 provider is staying in the middle of the call and not just
> performing a redirect to us.
> 
> I've tried the various CID settings in Asterisk, but am not able to use
> anything but our DID numbers for our outbound caller id.
> 
> My telco has been unresponsive to this issue.  
> 
> Does anyone know if it's possible with a PRI or do you have to have some
> other type of PSTN connection such as SS7?
> 
> Thanks!!
> --Shawn
> 

Shawn,


1)  When a call comes in, put the original CALLERID(number) into a 
variable.  This way, if you mess with the real CALLERIDNUM through your 
dialplan you can always set it back.  I like to use KKFROMCID to make 
sure that no scripts, Asterisk, etc mess with my original CID!

2)  Get a telco that lets you set any CID.  I don't know if I just look 
trustworthy or something, but I have had no problems whatsoever getting 
several LECs and CLECs in multiple states to let me set any CID I want. 
  Looking at the other posts, it seems that some people have problems 
with that.  I never considered it to be a big deal, just a cool 
privilege that you gain with a PRI...  It seems that isn't the case with 
some telcos.

3)  I don't know if this works or not, but I could swear that there is a 
redirect possible on PRI (similar to SIP 302).  I don't know if 
app_transfer (and your telco) support it, but it would be really cool 
because it would save you in terms of the number of used channels. 
(Using 0 channels instead of 2).

Check this thread:

http://lists.digium.com/pipermail/asterisk-users/2003-May/004594.html

--
Kristian Kielhofner