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Contributor II
Contributor II

Site To Site Extension Dialing

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Would it be possible to set up site to site extension dialing in the following scenario? In a private MPLS network, where Site A can ping Site B: Site A has a Netvanta 7100 that has SIP coming in from the carrier..  It is using 4 digit extensions. Site B has a Total Access 908e. Site B has a Toshiba CIX670 PBX.  The Toshiba CIX670 is not SIP capable at this time.  The 908e receives SIP from the carrier; there is a PRI link between the 908e and the Toshiba CIX670.  Site B is using 3 digit extensions. Would it be possible to have a SIP trunk configured in Site A 7100 pointing to Site B 908e and vice versa and somehow have Site A be able to dial Site B extensions and Site B dial Site A extensions?

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Contributor II
Contributor II

Re: Site To Site Extension Dialing

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After reading your post I asked staff at Site B to pickup a line same as when they will be dialing an outside call and dial Site A four digit extension.  To my surprise the PBX sent four digits over the PRI that I was able to process and route to the appropriate SIP Trunk.  So it is working Yay! Thx jayh, appreciate the help.

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Honored Contributor
Honored Contributor

Re: Site To Site Extension Dialing

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Yes. Build a SIP trunk between the two sites over the MPLS link. Work with your MPLS provider to tweak QoS as needed for call quality. Each end of the trunk points to the IP of the Adtran device at the other end. No need for registration.

To avoid dialplan conflict at site A, configure an unused leading digit (for example 7) to point to site B. Confugure a SIP grouped-trunk matching this and point it to site B. At site B accept the call and use match/substitute to strip the leading digit. The result should match the existing dialplan of the PRI. If the PRI expects more than 3 digits, then configure the match/substitute accordingly.

At site B, the Toshiba probably uses a leading 9 for outside calls as a trunk prefix.

Configure a second trunk with leading digit on the PBX, for example 8, with 4 digits to follow. PBX documentation may refer to this as a tie-line. Direct this to the PRI but don't strip the leading 8 as you do the 9. Set the SIP grouped-trunk at site B to accept 8XXXX and send to site A. At site A, use match/substitute to strip the leading 8 and it will match the extension dialplan.

Callers at site A dial 7XXX to ring site B. Callers at site B dial 8-XXXX to reach site A.

Substitute leading digits as needed if they conflict with existing dialplans.

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Contributor II
Contributor II

Re: Site To Site Extension Dialing

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Awesome.  Thx jayh.  I will attempt this.

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Contributor II
Contributor II

Re: Site To Site Extension Dialing

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Hi jayh, What if I already have SIP Trunks to other sites and are already using 2xxx, 3xxx, 4xxx, 5xxx, 6xxx, 7xxx, 8xxx? BTW, Site B uses 2xx and 3xx.

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Honored Contributor
Honored Contributor

Re: Site To Site Extension Dialing

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You may have to get creative. Site A doesn't likely use 9 for outside so could use 9xxx or 1xxx.

If any of your other SIP trunk locations don't use X[2-3]XX you could share the first digit in your dialplan. For example if SIP trunk 5XXX connects to a site which uses only 55xx through 59xx then you could share that one. 5[5-9]XX goes to the existing, 5[2-3]XX goes to the new one.

In the worst case someone has to renumber. Painful but they'll get over it.

If you're really using 2xxx through 8xxx efficiently, that's somewhere close to 7,000 extensions and probably about time to go to a 5-digit dialplan at site A.

A thorough dialplan analysis should result in the ability to do some consolidation.

If you do have to renumber something, plan for growth so that you only need to do it once.

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Contributor II
Contributor II

Re: Site To Site Extension Dialing

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Site A uses 9-xxx-xxxx for dialing outside local numbers if that is what you mean.  Are we able to use 1xxx? I was under the impression that we could not use 1xxx.  We are using [2-8][2-3]XX at 7 locations.

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Contributor II
Contributor II

Re: Site To Site Extension Dialing

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Update: I managed to allocate 82XX and pointed to the 908e at Site B. The call makes it to the 908e on the new SIP trunk but that is as far as it goes.  The PRI Trunk Group has a permit template set to $.  Do I need to add 2XX to the permit template on the 908e PRI Trunk Group?  Attached is output from SIP All Debug on the 908e.

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Contributor II
Contributor II

Re: Site To Site Extension Dialing

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Update: The call was coming in on the right SIP trunk but was then attempting to go out T25 which is SIP to the carrier.  On T25 I did a deny template for 82XX and the call then routed to the PRI T01.  The call still did not complete.  I'm now getting the following... Retrieving Data................................................................................................... 03:50:14:139 VOICE.SUMMARY T02 is calling T01 (8243). 03:50:14:140 VOICE.SUMMARY DNIS Substitution: dialed number 8243 -> 243 03:50:14:385 VOICE.SUMMARY Call from T02 to T01 (8243) ended by T01: unallocated number

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Honored Contributor
Honored Contributor

Re: Site To Site Extension Dialing

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It sounds like you're really close. You just need to do a little tweaking toward the PRI trunk at site B. It probably doesn't know how to handle 8XXX.

Are you sending the PRI 3 digits?  If so, on that trunk do the following, assuming the PRI is T01:

voice trunk T01

match 82XX substitute 2XX

match 83XX substitute 3XX

If it expects 4 digits with a different leading digit (for example 5)

voice trunk T01

match 82XX substitute 52XX

match 83XX substitute 53XX

If it expects 7 digits (for example 555-02XX) then do the following:

voice trunk T01

match 82XX substitute 555-02XX

match 83XX substitute 555-03XX

You get the idea, you want to match/substitute so that the PRI sees a digit pattern matching what it sees from telco trunks for DID to extensions.

And, yes on  the grouped trunk for the PRI accept 82XX and 83XX cost 0 but it sounds like you did that.

Nope, you said that you denied it on the other SIP trunk. That works too, but in my opinion a specific permit where you want it to go is a better solution than denying it everywhere else. Either way works, but if you add more trunks things may break until you go around and deny what you don't want on each one. Easier to specifically permit and possibly more secure in some scenarios. Most specific match wins.

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Honored Contributor
Honored Contributor

Re: Site To Site Extension Dialing

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"debug voice verbose" at site B would be more useful than the SIP debug. We know the INVITE is getting there. 🙂

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