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5 digit dialing to HQ internal extensions in SRST with DID

techguy73
Level 1
Level 1

I'd like to have the remote site (SiteB) able to dial 5 digits extensions of the HQ during the SRST mode.So from SiteB to route calls to HQ in srst mode, we need 11 digits handed off to the PRI which hits the HQ gateway delivering only 4 digits. Now, how to send any extra digits following the 11 digit sequence? There is no correlation between the last 4 digits and the DID number. In this example, say we only have one DID number, however on both sides (HQ and siteB) we have 20 phones. can we achieve this?

10 Replies 10

techguy73
Level 1
Level 1

I guess I might be answering my own question. So to support 20 extensions, we must have a DID block of 20 DNs from telco? The rest is all about translation rules and manipulating digits to be sent out in the last 4 digits of the whole 11 digit string. is this correct?

Hi

If you have a sensible dial plan where your internal extensions final digits match the external numbers you dial to reach them, then it's pretty easy to set up a dial-peer that matches the 5-digit internal number and manipulates that before sending to the PSTN.

If (as I believe is your case) your DDI numbers are randomly assigned to unrelated internal extensions, then you have more work. If there is no pattern, you can simply add a dial-peer for each extension at the HQ with a destination-pattern of the internal extension. As it's a POTS dial-peer, the entire matched string (5 digits) will be removed, so if you enter 'prefix

If there is some pattern to the number assignment you may not need a dial-peer for every extension; you'd have to work that out.

Regards

Aaron

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Aaron Please remember to rate helpful posts to identify useful responses, and mark 'Answered' if appropriate!

Dang, you typed just a little faster than me!  (Ok, a lot faster since you corrected my CME as SRST statement while I was working on this hahaha)

I have speed, if nothing else :-)

Aaron Please remember to rate helpful posts to identify useful responses, and mark 'Answered' if appropriate!

If you want to support 20 extensions that all need to be directly dialable from the outside world then yes, you'll need twenty unique extensions from your carrier.  A block of 20 contiguous extensions will greatly simplify your dial plan.

If getting 20 outside extensions is not possible or necessary you can use non-DIDs for your internal extensions.  Just mask all outbound calls to a main number and have a receptionist or auto-attendant transfer calls for you.

philip.e.denton
Level 3
Level 3

I'll answer each of your posts in order...

To enable SiteB to 5-digit-dial your headquarters you must translate the 5-digit extension the users are familiar with to the corresponding 11-digit DID coming in the headquarters PRI.  To do this, configure translation-rules and apply them to the proper outbound dial-peer (local or LD as appropriate).

If your site uses 5-digit extensions internally (rather than 10-digit extensions with 5-to-10-digit translations/expansions you'll also have do something similar to the calling number so the HQ users will see the proper number show up on the calling number info, missed calls log, etc.

The last time I checked translation-profiles could only hold 15 translation-rules at a time so if you're dealing with 15 random numbers and not a clean block of DIDs you may have to be pretty granular with outbound dial-peer/translation creation.

If your HQ site is already up and running there should already be correlations (translation patterns) on the inbound PRI to map the inbound 4 digits to whatever the actual directory numbers on the phones are.  Or is that not the case?

I'd agree with what you guys said, so for one to get the 20 phones rang over the PSTN, we need a block of 20 DIDs, if not then we are dealing with an AA type of setup to implement internally at HQ and Unity etc, however calling the branch under SRST would be a challenge if we don't have a block of 20 DiDs there, since establishing AA on an SRST router,,,hmmm not sure about that, I've done it for CME no problem but this isn't a CME.

Anyhow, lets say we are considering calling from SiteB to HQ with main number and AA, now the call can be sent to hit the main PRI but dealing with AA, now after the recording starts, introduces this Pause after which extension numbers can be accepted to ring them. Ok I can transform the 5 digits dialed by the person at SiteB to hit the PRI at HQ...BUT the challenge is how to send these digits after the 10/11 digit string is already completed to have the PSTN route that call back to HQ. Hope I'm not confusing you and you're getting my point. Is there a way i can introduce a Pause via translation rule. I know in old days we used to put "," for introducing a delay but it's not the case on cisco gw router.

P.S: In all other scenarios where there is enough DID numbers to cover for the number of extensions, the rest is just really to play with translation rules to implement that, whether dealing with simple dial plan or complex..(However you can ignore this scenario and discuss the one I presented in above paragraph).

According to the brief search I just did inserting a comma (,) into a dial-peer should introduce a pause in the digit sequence.  You'd end up needing one dial-peer for every number (block) you have but it should work.  Please let us know how it goes!

Thanks,

Phil

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so how would this dial-peer looks like.

Secnario Numbers.

HQ

1 PRI serving

Main DID number = 972-429-3444

Extension range at HQ = 52001 to 52020

Considering I only have one DID given above and not a Block.

AA is set up on 972-429-3400 putting people through to extensions 52001 to 52020

SiteB

1 PRI serving

Main DID number = 510-237-2333

Extension range at SiteB= 32061 to 32070

AA set up on above number putting people through to extensions 32061 to 32070

Now in SRST mode.

32061 is dialing 52011.

dial-peer voice 10 pots

destination-pattern 52...

prefix 19724293444,52

port 0/1/0:23

So the comma (,) there is going to work? if not where do we need to put it.

Hmm I'm not sure.  You've got a pretty strange issue here and I don't have a test environment where I could pilot it for you.  I think you're on the right track though.  Can you try this?

32061 is dialing 52011.

dial-peer voice 10 pots

destination-pattern 52...

prefix 19724293444,,,

forward-digits all

port 0/1/0:23

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