SIP and E1 Issues with 3rd party

[yamikani2g2]

Good day Experts

I have an issue i have a cisco router acting as a gateway with an E1 connection, then a CUCM they are connected to it using MGCP. I now have a SIP trunk to a third party device that is working, fine i can call the extension and they can call extensions on CUCM

1. Issue is i want 4 lines that are on the E1 to go onto the thirdparty pbx to extensions on it

2. I also want to call out Mypbx to the lines on the Cisco GW?

How can i achieve this?

[HARIS_HUSSAIN]

You cannot SPLT PRI with MGCP. You have to reconfigure to Use H323/SIP Protocol then we can split the Call on Voice Gateway

Below are the sample commands

controller E1 1/0/0
pri-group timeslots 1-31
trunk-group COMPANY-A timeslots 1-21
trunk-group COMPANY-B timeslots 22-31

dial-peer voice 1 pots
description *** FOR INCOMING to COMPANY-A ***
incoming called-number 4...
direct-inward-dial
trunkgroup COMPANY-A
!

dial-peer voice 2 pots
description *** FOR INCOMING TO COMPANY-B ***
incoming called-number 5...
direct-inward-dial
trunkgroup COMPANY-B
!
!
dial-peer voice 4 pots
description *** OUT TO PSTN COMPANY-B ***
destination-pattern 5T
direct-inward-dial
trunkgroup COMPANY-B
!
!
dial-peer voice 9 pots
description *** OUT TO PSTN COMPANY-A ***
destination-pattern 4T
direct-inward-dial
trunkgroup COMPANY-A

[yamikani2g2]

Please elaborate more how do i further push the calls for company B over the SIP trunk that i have between CUCM and Mypbx?

Should i maybe create a SIP trunk between CiscoGateway that has the E1 with mypbx? advise. meaning i will not have 2 SIP trunks

CUCM--------- Mypbx

CiscoGW-------Mypbx?

Is the above a better solution with the addition of your above solution.......

Lastly a twist if i split the E1 as said the numbers that have to go to company B  are random in the they are not in series.

[HARIS_HUSSAIN]

1) Incoming PSTN Calls
===> You can Create a SIP Trunk Between Voice Gateway and PABX to send incoming calls directly to the PABX.

 
dial-peer voice 11 voip
description *** TO CUCM FOR COMPANY-A ***
preference 1
destination-pattern 4...
incoming called-number .
voice-class codec 1
voice-class h323 1
session target ipv4:<IP>
dtmf-relay h245-alphanumeric
no vad
!
!
dial-peer voice 12 voip
description *** TO CUCM FOR COMPANY-B TO PABX ***
preference 1
destination-pattern 5...
incoming called-number .
voice-class codec 1
voice-class h323 1
session target ipv4:<IP oF PABX>
dtmf-relay h245-alphanumeric
no vad

2) Outgoing PSTN Calls from PABX Phones

             ==== Send the call Directly to Voice Gateway from PABX. 

3) Internal Call

              ====> For Call between CUCM Extension and PABX Extension We can use the SIP Trunk Between CUCM & PABX

Please rate and Mark Correct as applicable <===

[HARIS_HUSSAIN · Accepted solution]

To Answer you second query
Lastly a twist if i split the E1 as said the numbers that have to go to company B are random in the they are not in series

======> For Incoming call you need to create separate dial-peers  to appropriate Device.

For Example

If i have 30 Channel DID with Range 04 555 4100 -- 4300
So i need to allocates 4100 to 4199 to Company-A & Allocate 4200 ro 4299 to company-B
The Trunk Groups will only Split the Channels allocated for outbound calls. For Inbound we can create appropriate dial-peer to send call to respective CUCM/PABX depeding on the Dialled Number


dial-peer voice 11 voip
description *** TO CUCM FOR COMPANY-A 4100 to 4199***
preference 1
destination-pattern 41..
session-target ipv4:<CUCM IP ADDRESS>
!
!
dial-peer voice 12 voip
description *** TO CUCM FOR COMPANY-B TO PABX ***
preference 1
destination-pattern 42..
session-target ipv4:<PABX IP ADDRESS>

Please rate and Mark Correct as applicable <===