@SkykingOH - I did not mod Asterisk files directly, if that’s what you mean - I tried to find appropriate places in the FreePBX GUI wizards where to put what;
I shall also mention, I changed some things re. the default, e.g. changed into Users & Extensions mode [because some users will have more than 1 extension]
I have done a Trunk : (given it a descriptive name, then in the PEER Details:
fromuser=[my_SIP_username]
host=cust.ie.blueface.com
nat=yes
port=5060
qualify=no
secret=[my_secret]
type=peer
username=[my_SIP_username]
insecure=invite
then underneath
User Context : entered [my_SIP_username] into that one, then
in the USER details:
context=from-sip-external
host=cust.ie.blueface.com
type=user
In [my_PBX_address]/admin/config.php?display=sipsettings i tried changing NAT to different modes (mostly tried yes, no and route); canreinvite is set to NO;
Side note, my system spews some curious errors into log too :
[2014-07-23 09:10:34] ERROR[1823] pbx.c: You have to be kidding-- add exten '' to context app-blacklist? Figure out a name and call me back. Action ignored.
also this
[2014-07-23 09:10:41] WARNING[1878] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
and
[2014-07-23 09:10:42] WARNING[1823] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 1744 of /etc/asterisk/extensions_additional.conf
look curious…
As i mentioned, upon successful registration, I get this in CDR : (trimmed&anonymized)
2014-07-24 16:46:55 CHAN_START [caller] [caller] DEFAULT [trunk] from-sip-external SIP/[server_ip]-00000002
2014-07-24 16:46:55 ANSWER [caller] [caller] [caller] [trunk] DEFAULT s from-sip-external Answer SIP/[server_ip]-00000002
2014-07-24 16:47:03 HANGUP [caller] [caller] [caller] [trunk] DEFAULT h from-sip-external SIP/[server_ip]-00000002
2014-07-24 16:47:03 CHAN_END [caller] [caller] [caller] [trunk] DEFAULT h from-sip-external SIP/[server_ip]-00000002
2014-07-24 16:47:03 LINKEDID_END [caller] [caller] [caller] [trunk] DEFAULT h from-sip-external SIP/[server_ip]-00000002
Detailed Call Detail Records
2014-07-24 16:46:55 1406216815.2 [caller] Congestion s [from-sip-external] ANSWERED 00:08
but I only get it ONCE per PBX full restart, and after that, the providers’ server isn’t giving me any more calls, only logging attempts as ‘no answer’…
All I want for now is, that incoming calls start working reliably, hence I did not set up outgoing routes - incoming however is set, with destination set to ring all numbers (users) (for at least a test) - and I don’t know what am I actually missing…