SIP User not connected with Asterisk Server!

Hi All,

Soft phone not connecting with server,while giving all information properly.

Server Config details:

extensions.conf:

[general]

static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=yes

[phones]

exten => _X.,1,Dial(SIP/${EXTEN})
exten => _X.,n,Hangup()

exten => 111,1,Answer()
exten => 111,2,Playback(welcome)
exten => 111,3,Playback(demo-echotest)
exten => 111,4,Echo()
exten => 111,5,Playback(demo-echodone)
exten => 111,6,Playback(vm-goodbye)
exten => 111,7,Hangup()

sip.conf:

[general]

[1100]
type=friend
host=dynamic
username=1100
secret=abc1234
canreinvite=yes
nat=yes
context=phones
dtmfmode=rfc2833
disallow=all
allow=ulaw

Please suggest where i can modify to resolve the issue.

I see no issue so far. What is “not connecting” in SIP terms?

We’re going to need more information, such as what “not connecting” means like @AndrewZ asked as well as the output of “sip set debug on” with a call or register attempt.