How to define FreePBX user so I can log in with SIP softphone?

mholmes's picture

This seems a completely novice question, but nonetheless it has me stumped. I am merely trying to log into to my Asterisk server locally on my company's Intranet, using free SIP software (I've tried 3CX VoIP Phone and QuteCom). However, for the life of me I can't figure out where to define a USERNAME and PASSWORD for an Extension in the FreePBX administration interface, or in the various Asterisk .conf files.

I defined Extension 191 (config shown at bottom), and assigned it to an Incoming Route (which I doubt is relevant to merely connecting to the Extension). In the Device Options section of the Extension config page, there is a field for the password (secret), but not for a username/userid. I could not find any place in FreePBX to define an actual user, and all of the SIP software I've seen requires a username in addition to the password. What am I missing?

Many thanks to anyone who can be of assistance!

- Matt Holmes
mholmes@tmg1.net

Environment:
Dell PowerEdge 2850
Digium TE420 card
AsteriskNOW v1.5.0
CentOS Linux v5 final
FreePBX v2.5.1.0

/etc/asterisk/sip_additional.conf entry:
[191]
type=friend
secret=mypassword
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=191@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/191
context=from-internal
canreinvite=no
callgroup=
callerid=device <191>
accountcode=
call-limit=50


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191 is the username

AndrewZ's picture

191 is the username


Sorry, but no dice.

mholmes's picture

It sounded promising, but using "191" as the username did not get my SIP softphone connected to my FreePBX server.


(sorry for double post)

mholmes's picture

(sorry for double post)


Manually adding "username" to config didn't work either

mholmes's picture

FYI, I also tried manually adding "username=telnet191" to the /etc/asterisk/sip_additional.conf file and using "telnet191" as a username in my SIP software, but I am still unable to connect.


I'm able to connect now!

mholmes's picture

I'm not sure what I changed, but I am now able to connect to both extensions (191 and 190) with different SIP software using the extension number as the username, and I was able to make an internal SIP call from one extension to the other! Woohoo!

Now I just need to figure out how to get inbound and outbound calls working...