You will need user name and password from provider and type of connection. You need to change sip.conf on your server. In [general] section you should change this:

register => provider:qwerty@192.168.0.253/provider How SIP client will be registered on Asterisk
[provider] SIP client, you should this get from provider
type=peer type connection
host=192.168.0.253 IP address from SIP client
username=provider it should be same as name of SIP
fromuser=provider
fromdomain=192.168.0.254 Broadcast (probably) of provider
secret=qwerty password for auth
context=from-provider context in extensions.conf
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
canreinvite=no
nat=yes
insecure=invite
qualify=3600

In extensions.conf we have next configuration for dial plan [from-provider]. Default location for all ivr's is /var/lib/asterisk/sounds/ This is defined in /etc/asterisk/extensions.conf, as part of dial plan.

[from-provider]
include => Localcalls Dial plan includes some another dial plan, name for this is Local calls
include => ToAnotherOffice
exten => 1234567,1,Answer() in this point provider will sent call on 1234567 to Asterisk, Asterisk will answer
exten => 1234567,2,Background(en/ivr) in back ground will play IVR
exten => 1234567,n,WaitExten(5) Asterisk will wait for 5 sec to give caller chance to call some extension
exten => 1234567,n,Dial(SIP/600,15,tT) after that call will be routed to some extension and this extension will ring for 15 sec
exten => 0,1,Dial(SIP/600,15,tT) if caller press 0 in any moment he will be routed to extension
exten => 0,n,Dial(SIP/601,15,tT)