Here is my scenario. I want to have four (4) locations doing a SIP Trunk to PRI (Legacy PBX) handoff. The client just wants to do four (4) digit dialing between the sites using BroadBand and not the PSTN. What would be my set up scenario?
Does each location have a different extension prefix?
Meyerj66,
Thanks for posting! I believe you are working this issue through a normal support ticket. However, you can create three SIP trunks for each unit which specifies a SIP server as the corresponding far-end Adtran unit's IP address. Each of the SIP trunks would have a grouped trunk that specifies the extension range at that site. Below is a template for you to follow.
voice trunk T01 type sip
description Location2
sip-server primary <IP address of Adtran unit at this location>
!
voice grouped-trunk Location2
trunk T01
accept 2XXX cost 0
You can use that template for all your sites. Also, the for the PRI side of the unit, the following link has an example configuration.
Total Access 9XX - SIP to PRI sample configuration
Lastly, if SIP and RTP will be traversing a converged (VoIP and data IP traffic) network, just make sure that QoS will be implemented between the sites to prioritize DSCP 46 and ensure toll-quality voice.
Thanks!
David
Does each location have a different extension prefix?
Yes each location has a different extension prefix
Location1 - 1XXX
Location2 - 2XXX
Location3 - 3XXX
Location4 - 4XXXX
I forgot to add this is not an MPLS Network.
Meyerj66,
Thanks for posting! I believe you are working this issue through a normal support ticket. However, you can create three SIP trunks for each unit which specifies a SIP server as the corresponding far-end Adtran unit's IP address. Each of the SIP trunks would have a grouped trunk that specifies the extension range at that site. Below is a template for you to follow.
voice trunk T01 type sip
description Location2
sip-server primary <IP address of Adtran unit at this location>
!
voice grouped-trunk Location2
trunk T01
accept 2XXX cost 0
You can use that template for all your sites. Also, the for the PRI side of the unit, the following link has an example configuration.
Total Access 9XX - SIP to PRI sample configuration
Lastly, if SIP and RTP will be traversing a converged (VoIP and data IP traffic) network, just make sure that QoS will be implemented between the sites to prioritize DSCP 46 and ensure toll-quality voice.
Thanks!
David
Meyerj66,
I went ahead and flagged this post as "Assumed Answered". If any of the responses on this thread assisted you, please mark them as Correct or Helpful as the case may be with the applicable buttons. This will make them visible and help other members of the community find solutions more easily. If you still need assistance, I would be more than happy to continue working with you on this - just let me know in a reply.
Thanks!
David