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<div class="moz-cite-prefix">Definitely getting the caller id info -
see below. Its just ending up in the wrong field. The caller's
number is ending up in the "name" field, and the "number" field is
getting our G100's SIP peer name.</div>
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<div class="moz-cite-prefix">Its not clear if it is being offered
that way to the asterisk server that is accepting the call from
the G100, or if the asterisk server is mangling it before sending
it on to the customer... I can do some dialplan foo I suppose to
answer this question, but was really hoping someone would say "oh
I had that problem..." :)</div>
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<div class="moz-cite-prefix">j<br>
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<div class="moz-cite-prefix">On 2/15/19 12:45 PM, Nick Olsen wrote:<br>
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cite="mid:0eae142c0571409e80d06191ad0ded0c@floridavirtualsolutions.com">
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<div>You might confirm you're getting CallerID from the PRI in
the call setup. You can do a debug capture session on the G100
and get this info.<br>
<br>
If you need CallerID preserved from the PRI (Like the served
PBX sends multiple calling numbers based on end user station)
then you'll likely need to fix it on whatever the G100 is
serving with said PRI.<br>
<br>
If it's all one number anyways, You can just blanket overwrite
it from the G100 dialplan (I think it was in outbound routes).
Or ultimately, In the asterisk instance during receive before
shooting it upstream.<br>
<br>
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<div><strong>Nick Olsen</strong></div>
<div>Network Engineer</div>
<div>Office: 321-408-5000</div>
<div>Mobile: 321-794-0763<br>
<img
src="http://dl.floridavirtualsolutions.com/emaillogo/logo.png"
class="fr-fic fr-dii fr-fil" moz-do-not-send="true"></div>
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<hr id="previousmessagehr">
<div><span><strong>From</strong>: Jeff LaCoursiere
<a class="moz-txt-link-rfc2396E" href="mailto:jeff@stratustalk.com"><jeff@stratustalk.com></a><br>
<strong>Sent</strong>: 2/15/19 1:12 AM<br>
<strong>To</strong>: Asterisk Users Mailing List -
Non-Commercial Discussion
<a class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a><br>
<strong>Subject</strong>: [asterisk-users] Digium G100</span></div>
<div>Hi,</div>
<div><br>
</div>
<div>We recently dumped a Xorcom box that was no end of trouble
and replaced</div>
<div>with a Digium G100. PRI came right up, and we have been
using it fairly</div>
<div>flawlessly for several months now, with one caveat. Calls
that arrive</div>
<div>from the PRI are sent to the asterisk instance (13.23.1,
chan_sip), then</div>
<div>routed by the dialplan to various other gateways or
upstream providers. </div>
<div>When the call finally lands on a phone somewhere, the
caller ID</div>
<div>information has become corrupted, though in a predictable
way.</div>
<div><br>
</div>
<div>The CID number is replaced with the SIP trunk name of our
G100 gateway.</div>
<div><br>
</div>
<div>The CID name is replaced by the callers phone number.</div>
<div><br>
</div>
<div>This is problematic for a number of reasons - we have lost
the caller ID</div>
<div>name, if provided, completely. There is a lot of confusion
from our</div>
<div>customers asking "what does riisegw mean?!", and if they
try to return a</div>
<div>missed phone call or recall something from their history,
their phones</div>
<div>(Yealink models almost exclusively) try to dial to
"riisegw" since that</div>
<div>was actually in the number field.</div>
<div><br>
</div>
<div>I haven't tried to dig into this on our asterisk instance
yet, was</div>
<div>hoping this is something silly someone could direct us to,
or perhaps</div>
<div>someone from Digium can pitch in. I suppose I should have
some kind of</div>
<div>support with the G100... have never tried to actually call
Digium before.</div>
<div><br>
</div>
<div>Cheers,</div>
<div><br>
</div>
<div>Jeff LaCoursiere</div>
<div><br>
</div>
<div><br>
</div>
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