Limit calls on SIP trunk

[Phil Bradley]

We have a scenario where I would like to limit the number of simultaneous calls on a SIP trunk only in one direction based on a dialing pattern. This will be outbound calls coming from an IVR (Asterisk based) that I will pass to CUCM via a SIP trunk that will go out our T1. I can make the route pattern unique for these calls coming from the IVR such as start all with an 8 in our case. Then could I apply a rule to this pattern that only allows 3 calls at once for example? The purpose of this is we do not want to tie up our PRI with outbound calls from this application. These oubound calls could have hundreds of customers in the dial out program and I do not want this taking the entire pri circuit. I do understand that with another PRI I could route this pattern out a second pri but this is not an option yet. I have also considered routing these out a analog line but my TELCO does not have answer supervison on the FXO lines so the call out program starts playing the message as soon as CUCM opens the FXO port which is a problem.

[Thomas Dillingham · Accepted solution]

One way to accomplish this is on the Cisco IOS Gateway.

You will need an explecite incoming dial-peer that matches your pattern.

Then set:

max-conn 10 !!Your number of sessions here.

and

hunt-stop !!This will prevent it from hunting to the next best matching dial-peer

This should work to prevent too many calls from hitting the gateway at once.

TwD

[Phil Bradley]

Will this also work using MGCP from CUCM to the gateway?

[Thomas Dillingham]

No, you will have to have a SIP or h323 connection, MGCP does not allow this, and you won't be able to this with CUCM either.  So, h323 or SIP to the GW, is the only option to do this.

[Phil Bradley]

I was afraid of that. From what you are telling me max-conn is the only way to accomplish this? I have tried finding something on the SIP trunk itself but have been unsuccessful.

[Thomas Dillingham · Accepted solution]

You could use your router as a CUBE, by haripining your SIP calls through the CUBE and just send all SIP connections through the CUBE, then just translate incoming and outgoing to and from your PBX(s), thus creating a control point for all SIP.  You could even use flow-around and keep the media off of the gateway and just use the CUBE for call control. 

Just an idea. 

[Phil Bradley]

Thanks Thomas. I am not up to speed on CUBE but since that is an option I will mark your answer as correct. I have also learned that I can limit the active calls in Asterisk so I will probably use that option.