I am the IT admin for a private K12 school. We recently lost our old ESI phone system and I have been installing Asterisk based systems for over 10 year, so talked the school admins into letting me install the system to replace the ESI. Long story short, I have a new install running on a SIP trunk with 50 Grandstream phones. No issues with phones, users, VM, routing, etc… phone system is running perfectly.
The issue lies in the connection to my Valcom paging system. The Valcom system has a 2 wire interface for accept a call-in for the paging number. For instance, the old system would take the users input of 76208 and open the line to the Valcom 2924 and send 208 to it. Valcom has instructed me to connect a FXS gateway to the 2 wire input on the unit and set it up as a trunk.
So far, I have set up the HT503 FXS port with a username and password (using 701). I created a SIP trunk as below:
Trunk name: 701
Hide Caller ID
Max Channels 1
SIP Settings:
context=from-trunk
username=701
type=peer
secret=xxxxxxxxxxxxxxxxxxxxxxxxxx(password for extension from asterisk)
port=5060
host=(ip of HT503)
dtmfmode=rfc2833
Prefix Dial Patters
76 XXX
Outbound route is set to use this trunk with same dial patterns
When I dial 76xxx it just rings and the Valcom is not acknowledging. If I just dial 76, I get “all circuits are busy now”
The obihai is a whole other story…
Anyone have experience integrating a gateway into an analog paging system?
Thanks for any help!