Cisco Voice Gateway for Single Line

[TravisC]

I have a rather odd requirement. I have a single employee that resides in France (home worker). My voice system is a CUCM 11.5. This employee has a 8865 in MRA mode. I need to have this persons caller ID reflect his French phone number. So I ordered a DID from our SIP provider (voip.ms) and added a route pattern to route calls to France out this provider. So the calls go through BUT it appears the French Telecom is putting a +1 at the beginning of the caller ID due to the call originating from a US IP address. So far, France seems to be the only country doing this that I have encountered.

The end user does have a local POTS line (sorta) that we could utilize. So what I need is a device that I can register with CUCM that has an FXO port. Also, I don't want to overdo it as this is for a single user and a single POTS line. What device would best achieve this?

[John Steele]

VG400-2FXS/2FXO

https://www.cisco.com/c/en/us/products/collateral/unified-communications/vg-series-gateways/datasheet-c78-741973.html

This requires UCM 12 according to documentation.

VG310 supports FXO which works with 11.5

[Jonathan Schulenberg]

I don’t understand what’s wrong with the +1 prefix of the US DID; that’s the appropriate country code in E.164 and E.123 notation.

If you meant to say that you ordered a French DID from your provider, have you tried sending them a properly globalized calling/called number so they don’t have to guess?

There are also several design concerns here. If you are located in the US (an assumption) and the user is in Europe using MRA, the round trip delay across the Atlantic and back will destroy the Q.850 delay budget, rendering the call unusable. The only way this would be even remotely viable would be a router with VPN and the phone connected to that router. That way calls over the local FXO port could just route traffic to the local phone. Signaling would still have to backhaul and while there may be some sluggishness or audio cut-through delay, at least the call audio would be OK once connected. Just be careful about analog loop disconnect supervision.