I am having a hard time wrapping my head around this. I would like to send just a string of DTMF Tones out a fxo gateway.
I am having trouble setting up the trunk to the gateway, The only way I could get the trunk to connect was to to create an extension and put in the extension creds. I am not sure what to use for a username and password to the FX0 Gateway itself.
Thanks @stewart1, I wanted to reply to my previous but it auto closed before I got the adapter.
I read through that and It looks like he was saying he was getting a 403 forbidden back from the HT503. When I do a TCP dump I am seeing all 200OK and when I go under asterisk info in the chan_sip section I see this, which shows me that it is okay?
I can’t explain the 995161 but it’s likely not relevant to the FXO issue. You could try fwconsole restart
and maybe it will go away, or try grep 995161 /etc/asterisk/*conf
and see if anything appears that you could track down.
Anyway, at the Asterisk command prompt, issue sip set debug on
and make a call attempt.
Paste the relevant section of the Asterisk log at pastebin.freepbx.org and post the link here.
I had also previously set debug peer on, sip set debug on probably just overrode it. I wasn’t 100% which part was the relevant part. So I put the whole thing
[2019-08-13 11:52:58] VERBOSE[7124][C-00000222] app_dial.c: Called SIP/BHSPaging/#10@BHS_Paging
Where did you use the name BHS_Paging (with the underscore)? That will cause trouble. The OP in the referenced thread had a similar issue that went away after he set the trunk name and username to be the same and did an fwconsole restart.
Also, the Dial Plan in the HT won’t accept the # character by default, so edit that to e.g. {#x+|x+}
and also set Use # as Dial Key to No.
The name of the trunk is BHS_Paging I renamed to be the same as the username, for the Dial plan do i change that in the ATA and leave the trunk dial plan to “System” in the GUI?