connection tie-line question

[ciscoforum]

HQ PBX has two E&M T1 CAS, connecting to GW's two T1 CAS ports. DID numbers for HQ is not seperated between the CAS ports. There are two remote sites with GW connecting to its local PBX. Now my question is: Is it possible that using connection tie-line can make sure call from Rs1 to HQ always use T1 port 1 on HQ, and call from RS2 to use T1 port 2 on HQ. Remember HQ DID number are in one pool. Dial peer on RS are same. Thanks. Somebody mentioned to me that tie line is like a tunnel that you can define to use which port on the other end for outbound. But it seems that it doesn't work that way.

[ciscoforum]

Thanks very much. I am one more step further now.

Now on the above scenario, I have to add 1 PRI on the HQ and add one more Branch, say BR5 which it only has PRI. Understand connection trunk not work on PRI, that's fine. I don't need that.

The requirement is: Make sure call from BR5 has to use this PRI only.

My question is: when I have connection trunk configured to exclusively for the other 4 T1 CAS sites, I still should be able to use regular dial-peer to route the call from BR5 to use PRI to PBX based on the called number. To me, these are two different process, the router should be able to identify without confusion ,right?

2. On the previous thread you mentioned there is another way to send call on specific ports. I will be more than interested in hearing that. All I know that router has to route the call based on the called number in regular way.

[paolo bevilacqua · Accepted solution]

The router will make no confusion as long the numbers used in dial-peer are correctly configured. Eg, if you have a mix of "router private" and "PBX dialplan" numbers, these do not overlap.

If you like the idea of using "connection trunk" tecnique with PRI too, it is possible to do that as well. It is called t-ccs (transparent CCS). Signaling timeslot is transported transparently with codec tccs, and voice channels can be compressed and VAD applied as desired.

Personally, unless proprietary ISDN IEs are used and the router is unable to pass them, I see no advantage in doing that.

The links about sending calls coming from VoIP to specific POTS, based on calling number:

http://cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801c0a88.shtml

http://cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml

[ciscoforum]

This link is very good. But we may not able to do the calling number translation on the tieline(E&M), if it does not support calling number on the signaling. I know only problably group D supports it.

[paolo bevilacqua]

For "connection trunk" you never need to worry about matching on calling number, because the destination number is unique, and will match one destination port only.

However, if you wanted to insert and arbitrary calling number on ports that do not have it, you can use "station number" command for FXS, or a "translate called" for any type of port.

Calling number (ANI) is supported in E&M Feature Group B. Another variant of CAS that supports ANI is Feature Group D, that is used on some PSTN trunks.