I have my asterisk system set up where I can receive and place external phone calls through a Grandstream ht503 on a pstn line. I can dial out with not problem, but when I dial in, I get an internal dial tone, at that point I can then dial an extension number and it will ring that phone instead of going straight to an IVR that I have recorded and set up on the server. I’ve searched the internet for information on this, but have found nothing. I am new to asterisk and so far this is the only issue I’ve had. Any help or suggestions are appreciated. Thanks!
What is your inbound route pointing to? Almost sounds like you have it pointing to a DISA.
BF
I would venture a guess that the dial tone is coming from the Grandstream, not asterisk. You could log into the CLI to watch and see at what point asterisk gets the call.
My inbound route destination is pointing to my IVR, the did number is 500 which is the sip extension for the ht503. I’m new to asterisk, how do I log into the CLI?
Thank you all for your feed back, I got it working. It was a setting on the ht503. I had to create misc application on asterisk that would create a dialable number for the ivr (in this case I used 7777 as the number). On config for ht503 under “basic settings” all the way at the bottom there is a section called “unconditional call forward to VOIP”. The user id was empty so I typed in 7777 and applied the settings and everything works now!