Asterisk: Connecting an Asterisk System To SIP Provider - asterisk

Question

Setup:
Centos 6
OS: Linux CentOS 64-Bit
CPU: Intel® Core™ i7 - 4 cores
Asterisk 11
libpri 1.4
dahdi 2.9.1
I want to send calls to my SIP provider via asterisk. I want to use a softphone to make outgoing calls, when I make outgoing calls on the softphone it needs to route through my asterisk server and then out to the SIP Provider. Currently when I make an outbound call it produces a "Failed to authenticate" and status is 'CONGESTION' notices.
My question is simple: Since my softphone is calling from "User1" (as shown below) What do I need to write in my sip.conf and extensions.conf files in order for the SIP provider to make the outgoing call for me?
My Thoughts:
I feel like I am missing a part of the process, like how User1 is set up to handle outgoing calls...
See an overview of how I set these two files up currently:
notes: - all username and passwords have been removed for this post. - sip.us is the sip provider
sip.conf:
[general]
transport=udp
register => username:password#gw1.sip.us
[myprovider]
type=peer
insecure=port,invite
host=gw1.sip.us
port=5060
dtmfmode=rfc2833
canrevinvite=no
disallow=all
allow=ulaw
qualify=yes
qualifyfreq=30
nat=force_rport,comedia
trustrpid=yes
fromdomain=gw1.sip.us
username=*****
secret=*****
context=from-trunk
rfc2833compensate=yes
session-timers=refuse
[friends_internal](!)
type=friend
host=dynamic
context=users
disallow=all
allow=ulaw
allow=alaw
[user1](friends_internal)
secret=****
[user2](friends_internal)
secret=****
now the extensions.conf:
...
...
exten=>_1NXXNXXXXXX,1,Dial(SIP/${EXTEN}#gw1.sip.us)
[users]
exten=>6001,1,Dial(SIP/user1,20)
exten=>6002,1,Dial(SIP/user2,20)
now the asterisk cli output when i try making an outgoing call using softphone:
== Using SIP RTP CoS mark 5
-- Executing [12015550155#users:1] Dial("SIP/user1-0000001e",
"SIP/12015550155#gw1.sip.us") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/12015550155#gw1.sip.us
[May 4 10:02:13] NOTICE[28935][C-0000002a]: chan_sip.c:23094
handle_response_invite: Failed to authenticate on INVITE to
'<sip:user1#501.62.174.2>;tag=as798e5009'
-- SIP/gw1.sip.us-0000001f is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/user1-0000001e' status is 'CONGESTION'
asterisk cli> sip show registery
Host dnsmgr Username Refresh State Reg.Time
gw1.sip.us:5060 N username 105 Registered Sun, 04 May 2014 10:28:05
1 SIP registrations.
asterisk cli> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status
user1/user1 68.198.. D Auto (No) No 55461 Unmonitored
user2/user2 68.198.. D Auto (No) No 55461 Unmonitored
myprovider/username 65.254.44.194 Yes Yes 5060 OK (42 ms)

Answer 1

Try changing the #gw1.sip.us to #myprovider and see if there's any changes. This will force asterisk to use the configured trunk [myprovider] instead of directly forward the call to gw.sip.us directly.
Hope this helps.

Answer 2

Your DialPlan is not correct clearly from your configuration files.
Аs a first step change your register string like:
register => username:password#gw1.sip.us\Myprovider
and then add the outgoing and incoming dialplan in your extensions.conf and include these dialplans into your users context. like:
e.g: you use 6XXX series to dial to the provider:
[outgoing]
exten => _6XXX,1,Dial(SIP/Myprovider/${EXTEN:0})
exten => _6XXX,2,Hangup
and for incoming calls
[incoming]
include = users ; this will go into your local context and dial your extensions
and in users context add outgoing context like
[users]
include = outgoing
I hope this will help you :)