Connecting 2 or more TA900 Series to one SIP trunk account

[Anonymous]

I have a need to connect several TA900 series to one SIP trunk account.  I have 62 FXS lines that I need to connect to one SIP trunk group.  I have 23 call paths on a trunk account from my provider but I don't have enough ports on any one unit.  I need to register one device to my provider and register the other devices (Slaves) to the first one (Master).  I have been able to get the FXS ports on the Master to work...This is a registered trunk, the DID's don't register with the provider.  The Master device is a TA908e (3rd Gen) and the Slaves are TA924's (2nd Gen).  Any help would be greatly appreciated!

[jayh]

Does it have to stay SIP?

Configure three T1s on the Gen 3 as PRI role network, connect them via T1 crossover cables to the T1 0/2 ports on the TA924s configured as PRI role user. Configure a grouped-trunk for each PRI on the 908e with the DIDs going to its respective 924.  Done!

You'll oversubscribe the DSPs on the TA908 but with only 23 call paths you'll have plenty to spare. 

[Anonymous]

The calls will route over Ethernet ports, and the 3 Slaves are not in the same building but on the same internal subnet.

[jayh]

OK, you should be able to make this work.  Create a SIP trunk on the master for each slave, pointing to the private IP of the slave.  Create a grouped-trunk for the group of DIDs facing each slave on the master.  Enable media-gateway ip primary on all Ethernet interfaces participating in SIP.  I don't think you'll need to do any SIP registration, just use the internal IPs of the slaves as the SIP trunk destinations.  If NAT is involved you may need to enable SIP proxy on the master or possibly get the SBC feature set (but I don't think so). This depends somewhat on how well your provider handles NAT traversal.

[Anonymous]

I have the provider trunk account up and registered, so I think I am good there.  I'm a bit confused on the grouped-trunk for the group of DIDs facing each Slave.  The DID's are random so I'm not sure exactly what you mean.  What about on the Slave, will I create a trunk account with the Master unit IP or since my 908 is an e version would I need to set up a separate IP there for talking back and forth to the Slaves?  Thanks for the Help!!!

Here is what I have so far:

voice trunk T02 type sip
  description "Master to Slave"
  sip-server primary 10.26.66.24
  grammar request-uri host domain
  grammar from host domain
  grammar to host domain
!
!
voice grouped-trunk Test
  trunk T01
  trunk T02
  accept NXX-XXXX cost 0
  accept 1-NXX-NXX-XXXX cost 0
  accept 1-800-NXX-XXXX cost 0
  accept 1-888-NXX-XXXX cost 0
  accept 1-877-NXX-XXXX cost 0
  accept 1-866-NXX-XXXX cost 0
  accept 1-855-NXX-XXXX cost 0
  accept 011-$ cost 0
  accept 411 cost 0
  accept 611 cost 0
  accept 911 cost 0
  accept 0-NXX-NXX-XXXX cost 0
  reject 976-XXXX
  reject 1-900-NXX-XXXX
  reject 1-976-NXX-XXXX

[jayh · Accepted solution]

I'm not sure what you mean by "The DIDs are random".  My understanding is that there are 62 telephone numbers (DIDs) being delivered on a SIP trunk with 23 simultaneous calls allowed and that each telephone number is to ring on a specific FXS port on a specific chassis.  So, for example:

312-555-0101 through 555-0108 go to the 8 FXS ports on the 808e.

312-555-0109 through 555-0132 go to the 24 FXS ports on the first 924.

312-555-0133 through 555-0156 go to the 24 FXS ports on the second 924.

312-555-0157 through 555-0162 go to six of the FXS ports on the second 924.

Only 23 simultaneous calls are allowed.

If this isn't similar to your scenario, what are you trying to do?

Anyway, on the config, pretend that the number plan above is right.  On the master:

voice trunk T01 type sip

description "SIP Provider trunk"

sip-server primary [IP or hostname of provider]

!... rest of existing working provider trunk config.

!

voice trunk T02 type sip

  description "Master to Slave No. 1"

  sip-server primary [IP of Slave 1]

!

voice trunk T03 type sip

  description "Master to Slave No. 2"

  sip-server primary [IP of Slave 2]

!

voice trunk T04 type sip

  description "Master to Slave No. 3"

  sip-server primary [IP of Slave 3]

!

voice grouped-trunk SIP-Provider

! [You probably already have something like this]

  trunk T01

  accept NXX-XXXX cost 0

  accept 1-NXX-NXX-XXXX cost 0

  accept 1-800-NXX-XXXX cost 0

  accept 1-888-NXX-XXXX cost 0

  accept 1-877-NXX-XXXX cost 0

  accept 1-866-NXX-XXXX cost 0

  accept 1-855-NXX-XXXX cost 0

  accept 011-$ cost 0

  accept 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept 0-NXX-NXX-XXXX cost 0

  reject 976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

!

voice grouped-trunk slave1

trunk T02

accept 3125550109

accept 31255501[12]X

accept 312555013[012]

!

voice grouped-trunk slave2

trunk T03

accept 312555013[3-9]

accept 312555014X

accept 312555015[0-6]

!

voice grouped-trunk slave3

trunk T04

accept 312555015[789]

accept 312555016[012]

!

!

voice user 3125550101

connect fxs 01

!... and so on

On each slave, configure the voice users to the FXS interfaces, create a voice trunk T01 type sip pointing to the inside IP of the master, and a grouped-trunk allowing all to go to it.

If there are a lot of extension-to-extension calls you could build individual trunks and grouped-trunks to fully mesh the slaves but they'll hairpin through the master OK.

[Anonymous]

Hello,

I just wanted to check back in with you to see if you are still having problems.  If so, feel free to respond with any additional questions you may have

here.

Regards,

Geoff

[Anonymous]

Hello Daniel,

I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily. If you don't feel like the answer I marked was correct, feel free to come back to this post and unmark it and select another in its place with the applicable buttons.  If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.

Thanks,

Geoff