Digium G100

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Asterisk Users 2 Comments

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

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2 thoughts on - Digium G100

  • 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.

    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.

    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.

    Nick Olsen Network Engineer Office: 321-408-5000
    Mobile: 321-794-0763

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  • 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.

    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…” 🙂

    j