Incoming Call classification on single trunk

Unanswered Question
Mar 11th, 2010
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Is there a way to provide CAC bandwidth control of incoming calls on one SIP trunk and to be separated per classification by configuring CCM7 alone?

For example:
PSTNA ----sip trunk---CCM7----Phone1A

Phone1A and phone1B are in location san jose. Phone 2A and 2B are in location RTP.
I want to limit the incoming call on San jose location to 80kbps g711 single call only. While on RTP location, i want to limit 160kbps g711 incoming call.
If i configure 2 sip trunk with the same ip address but different trunk name, CCM will accept it and i can configure different locations corresponding to each trunk. But how can i instruct the incoming call to use the trunk for San jose and the other to use RTP? based on called number maybe?

I can apply CSS on the trunk to know which phones but the incoming call needs to know which sip trunk to use first. If the sip trunk is a separate ip,then it would be simple but in my case, it is only one trunk.

Is there a way? How to do it?

I hope my explaination is clear.


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Overall Rating: 5 (2 ratings)
Aaron Harrison Fri, 03/12/2010 - 01:20
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In short, you can't... an incoming SIP message is linked to a trunk by the IP/port pair, so if both messages come in from the same IP to the same port, they are both the same. You can't differentiate based on the number in the INVITE.

If you only allow 80K to one site, and 160 to another, why not just put phone 1A/1B in one Location, and Phone 2A/2B in another location, and set 80/160K on those two locations?


jechoi Fri, 03/12/2010 - 03:24
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Hi Aaron,

Our goal is actually to restrict the number of incoming calls that is routed to each ip phone on each branch.

The provider told us that they do not support restricting number of calls sending to each site and were hoping CCM could actually workaround for it Problem is all each branch site is registered to a single CCM.

I see. So not possible then.

Thanks for the help.



Aaron Harrison Fri, 03/12/2010 - 03:28
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It IS possible, it's just not a property of the SIP trunk.

You would group the phones into 'Locations', and set each location to a suitable bandwidth.

e.g.. phones on site A go into location 'Site A', set to 160K

phones in site B go into location 'Site B', set to 80K.

Leave the SIP trunk in hub_none.

When a call comes from the SIP trunk to Site A, running G711, it takes up 80K of the available bandwidth. The second call would take the other 80K, and the third call would be rejected.

For Site B, a second call would be rejected...



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