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FXO card, catch DNIS & ANI? translation-rule use?

xiaoliangyue
Level 1
Level 1

Hi, gurus,

I've got a router 2801, with 6 FXO ports; the router runs CME.

1) Is it possible for the Telco to send both DNIS & ANI to the router?

2) If the router catchs DNIS, can I use translation-rule to transfer a call to a specific extension?

Thanks.

5 Replies 5

paolo bevilacqua
Hall of Fame
Hall of Fame

Hi,

DNIS and ANI are avaiable only on sigital interfaces, not analog lones.

On analog lines, you can have only DID (trunk lines- that will signal the called extension.

Thanks for reply.

As I understand, if router is equipped with DID card, the router can receive digits from Telco, and transfer the call to specific extension. This's how DID works? in this case, is the analog line a trunk? I'm a bit confused in here. Thanks.

All FXO ports terminate standard analog lines, which are often referred to as "1FB"s The analog line does NOT pass ANI or DNIS because it is a simple signaling wire pair and does not have a mechanism for "reading / storing / forwarding" so to speak the Caller ID messages produced by the CO switch. This is often why you need a Caller ID enabled handset or a smartbox to interpret the data passed. A digital circuit, such as a PRI, are able to track caller ID by use of what are called progress messages, which allows the ANI to be embedded and passed to the CPE and used for queueing to a sig digit extension or "main number line" if you will. Typically, the best workaround for an FXO is to just use a plar mechanism or a null voice translation rule such as:

voice translation-rule

rule 1 ^$ 1000

voice translation-profile AA

translate called 1

dial-peer voice 1 voip

destination-profile incoming AA

** this takes any inbound calls on the FXO and converts the null digit string to 1000 and allows you to PLAR the call off to an FXS or a hunt group.

I know some of my telco jargon is probably not EXACT, but this is the general idea...

HTH,

rhugo,

what you said is correct, only I'm not sure why you suggest the complication of translation rules etc for a thing that as you mention, is doen with a single 'connection plar' command.

Also, the exaple you have give, does not work in practice. First, the trnalation rule must be applied to a either a voice-port or pots incoming dial-peer. But in the example you made, threre is a voip dial-peer, and a 'destination-profile' command that doesn't exist.

Thanks, guys.