I have a CME 8.6 system with approximately 50 sip phones and about 10 skinny. Most of the phones are programmed to send the outbound caller id of the main phone number with a generic ... translation. (I can post configs if necessary). A few phones send their DID numbers and that seems to work.
My question is, I have a few sip end points (mixed bag of phones, mostly very low end) that send out anonymous caller id even though my translations are the same for all phones.
System is connected to a Comcast PRI.
Where can I look to find this weirdness?
Considering contacting Comcast and just having them override any anonymous caller id from our PRI, but I feel like that is a last resort, or is it?
Thanks for educated advice.
Solved! Go to Solution.
I recommend running DNA (Dialed Number Analyzer) using the source phones with the issue with one of the outside phones seeing the wrong caller ID. See what DNA shows as the source of the anonymous.
Also, if you haven't already, double check from the gateway that the calls are being sent to Comcast as anonymous and that Comcast isn't setting them to anonymous. Could save you a lot of work. :-)
This answer is the ticket:
Also run a "test voice translation-rule <master tag> <DN>" in exec mode to see if the translation take affect for your DN.
I ran that command and I can see I'm sending the right information to the provider. I'm calling the provider now.
Of course - missed that it was CME. Since it's somewhat on-topic, I thought it would be interesting and helpful to mention that CUBE does have DNA of a sort (see link below). It may not be helpful in this case, but could be helpful elsewhere.
I thought I had this issue solved but it turns out to be slightly different that I thought originally.
On an older CME system, I have about 10 ephones and 50 or so low-end sip phones.
All of the SIP phones generate restricted for caller-id when calling out across the PRI (names work internally). The ephones work with proper caller-id.
I've done a couple of ccsip inout debugs for a working and non-working extension (ephone and sip extension), and what I see is the ephone grabbing an ambiguous dial-peer of 42003 and then hitting outgoing dial-peer 62 in the attached config. For the SIP endpoint, it goes straight to the dial-peer 62.
I'll attache the debugs next.
I know someone smarter than I can figure this out :)