[asterisk-users] Digium G100

[Jeff LaCoursiere]

Hi,

We recently dumped a Xorcom box that was no end of trouble and replaced 
with a Digium G100.  PRI came right up, and we have been using it fairly 
flawlessly for several months now, with one caveat.  Calls that arrive 
from the PRI are sent to the asterisk instance (13.23.1, chan_sip), then 
routed by the dialplan to various other gateways or upstream providers.  
When the call finally lands on a phone somewhere, the caller ID 
information has become corrupted, though in a predictable way.

The CID number is replaced with the SIP trunk name of our G100 gateway.

The CID name is replaced by the callers phone number.

This is problematic for a number of reasons - we have lost the caller ID 
name, if provided, completely.  There is a lot of confusion from our 
customers asking "what does riisegw mean?!", and if they try to return a 
missed phone call or recall something from their history, their phones 
(Yealink models almost exclusively) try to dial to "riisegw" since that 
was actually in the number field.

I haven't tried to dig into this on our asterisk instance yet, was 
hoping this is something silly someone could direct us to, or perhaps 
someone from Digium can pitch in.  I suppose I should have some kind of 
support with the G100... have never tried to actually call Digium before.

Cheers,

Jeff LaCoursiere