The Adtran community holiday season is starting next week! The holiday period will span from December 21, 2024 to January 6, 2025. During this time, responses to feedback form submissions may be delayed. If you are encountering product issues, you can reach out to Adtran support at any time.
cancel
Showing results for 
Show  only  | Search instead for 
Did you mean: 
Anonymous
Not applicable

Limiting incoming calls on SIP to PRI TA908 3rd Gen

All,

So I have a current customer that's using a legacy PRI and I need to migrate them to our current SIP to PRI platform. In the legacy setup they're able to limit the incoming calls across their PRI so incoming doesn't ever use the entire PRI. Is there a way to limit the amount of incoming calls so say 10 calls can be incoming leaving 13 open channels for outbound calls? Thanks for any ideas.

0 Kudos
1 Reply
Anonymous
Not applicable

Re: Limiting incoming calls on SIP to PRI TA908 3rd Gen

Can you limit inbound calls from the PBX itself? It wouldn't matter who provides the PRI in that case.

To do this on the TA908 I see three possible solutions, one on the SIP side and two on the PRI.

On the SIP side, in your voice trunk type sip, you could try 'max-number-calls'. Create one trunk for inbound SIP and another for outbound SIP, apply the max to the inbound trunk only. Set dial plan to/from the PRI accordingly. This method allows either side of the PRI to seize any open channel.

On the PRI side, you could use separate isdn-group for inbound and outbound. Use 'max-channels 10' on one for inbound, and maybe 'min-channels 13' on the other for good measure. Then create two voice trunk type isdn, one for inbound and one for outbound, and 'connect isdn-group' to the appropriate group. Set the dial plan to/from the SIP trunk accordingly. This method also allows either side of the PRI to seize any open channel.

The other PRI method starts at the underlying T1 interface: create two tdm-group using different timeslots, e.g 1-11 and 12-24. Then create two separate PRI interfaces using "connect t1 ..." to each tdm-group, and two isdn-group and two voice trunk type isdn per the above. Set the dial plan to/from the SIP trunk accordingly. This method basically gives you two fractional T1 at the cost an additional D-channel, and the two trunk groups also needs to be configured on the PBX so it will seize the correct channels if it will even support this. I don't know why anyone would want to do it this way.