[chawlakunal]

Hi,

I am trying to register my Asterisk as a Client at a SIP provider which provides PSTN access so that I can dail in and out on PSTN using SIP softphone (X-Lite). Now, I am able to register Asterisk against the SIP provider and get incoming calls on softphone too. But the problem is with outgoing calls. After dailing the PSTN number the PSTN phone rings but even after picking the PSTN phone the softphone displays calling 0xxxxxxxxxx (PSTN number). Then finally the sip softphone displays "Call Failed: Service Unavailable" and you hear the voice "The person you called is unavailable".

The Settings in sip.conf are:

general
port = 5060
bindaddr = 0.0.0.0
context = others
sipdebug = no
realm = domain.com
trustrpid = yes
sendrpid = yes

register => uname@domain.com:pwd:authname@IP/46
registertimeout=20
registerattempts=10


my_provider
type=peer
fromuser=uname
fromdomain=domain.com
canreinvite=no
secret=pwd
insecure=very
host= ip
qualify=yes
nat=no


The configuration in extensions.conf is as follows:
exten => _0.,1,Dial(SIP/${EXTEN:1}@my_provider)

The output on Asterisk CLI is:

Executing 04045834323@tutorial:1 Dial("SIP/alice-c0000a60", "SIP/4045834323@my_provider") in new stack
— Called 4045834323@my_provider
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/alice-c0000a60' status is 'CHANUNAVAIL'


Can someone please explain where and what I am doing wrong?

Thanks.