Sip trunk to ITSP load balance

[Alan Couto]

Hello guys,

Client has a PABX and will use a Cisco2911 with E1 and will have Sip Trunk to a ITSP.

With that client want to load balance some channels from E1 to each interface of the router. Example: channel 1 to 20 goes through GE0/0 and 21-30 through GE0/1.          

Is that possible? If not there are any similar way?

Thanks,

Alan.

[Ayodeji Okanlawon]

Alan,

What do you mean by channel 1-20 goes to ge0/1?? Can you explain better??

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Alan Couto]

Hello,

Thanks for reply.. I not usually works with post sales/implemantation but let me try.

Client will have a connection from 2911 to PABX with E1, so I imagine that we will have some channels group from 2911 to PABX. Client want to separate this channels groups like channel-group 0 timeslots 1-20 and channel-group 0 timeslots 21-30. I think this is the easy part.

The second part, client will have a SIP trunk from 2911 to ITSP and want to make load balance. He wants that some calls go through one interface and another calls goes through another interface. With that, if one link goes down some extensions will work.

I was looking at google at bind commands that we can make like a source interface.. working together with 2 voip dial peers with a incoming number x... and incomming number y... but dont know if make sense.

The other way that I thought is use the channel-group and try to point to a dial-peer voip with bind commands but also dont really know if is possible.

Thanks!!

[Ayodeji Okanlawon]

Alan,

Is this solution going to use two different ITSPs? If that is the case then why are you not using two different gateways..If thats not the case then what type of redundancy do you think this will give? Are you going to have two differnet IP addresses from the ITSP to configure on the two different interfaces?

This doesnt look like a well designed solution. I dont see how you can have redundancy based on two interfaces non a router..If the router dies, then there goes your redundancy...

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[Alan Couto]

Hello aokanlawon,

Thanks for the attention. I called today to the client to understand a little bit more the scenario.

Client told me that actually they have two different offices very close to each other and they use a shared PABX at same location.

Client will install a gateway with E1 to PABX and the channels will be separated like that:

DDR 1500 a 1549 channels 1 to 20 - OFFICE 1

DDR 1550 a 1599 channels 21 - 30 - OFFICE 2

And will have a Sip Trunk with ITSP but they want to use one Giga interface for SIP trunk 1 and other Giga interface for SIP trunk 2 where:

Sip trunk 1 will have to deliver DDR 1500 a 1549 to channels 1 to 20 and

Sip trunk 2 will have to deliver DDR 1550 a 1599 to channels 21 - 30.

This make sense? How can we translate it in a configuration? Does we know any example?

Many thanks!!

[Ayodeji Okanlawon]

First of all, I dont see this as a feasible solution. I doubt if an ITSP will give you one SIP trunk and split the DDI as you are proposing...The implication of then doing this is that they will need to give you two seperate ip addresses etc...You must speak with them first to see if they can do this...

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Alan Couto]

Hello aokanlawon,

I talked with the ITSP guy who is selling the solution and he said that provide two Ip address will not be a problem it was already at the plain..

[Ayodeji Okanlawon]

OK in that case here is what needs ti be done.

1. define two trunk groups on the gateway

trunk group GROUP1

description ****  DDI RANGE 1500-1549*****

hunt-scheme sequential

trunk group GROUP2

description **** DDI RANGE 1550-1559*****

hunt-scheme seqential

2. Seperate your E1 channels and assign trunk groups to them

controller E1 0/0/0

pri-group timeslots 1-31

trunk-group GROUP1 timeslots 1-20

trunk-group GROUP2 timeslots 21-30

3. Configure an inbound sip dial-peer for your DDI range

dial-peer voice 1 voip

incoming called number 15..

session protocol sipv2

dtmf-relay rtp-nte

no vad

4. Configure your pots dial-peer for the first range and assign the trunk group to it

dial-peer voice 2 pots

trunkgroup GROUP1

description DDI Range 1500-1549

destination-pattern 15[0-4].

progress_ind setup enable 3

progress_ind alert enable 8

forward-digits all 

5. Configure the second pots dial-peer and assign the second trunk group

dial-peer voice 3 pots

trunkgroup GROUP2

description DDI Range 1550-1559

destination-pattern 15[5-9].

progress_ind setup enable 3

progress_ind alert enable 8

forward-digits all 

NB: This solution will work for incoming calls from the ITSP to your gateway.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Alan Couto]

many thanks aokanlawon, I will send this information to client and will let u know.