Asterisk Outbond call issue

Asterisk Asterisk Support
1

i’ve flowroute,voxbeam configured SIP server and its working fine .
Yesterday 've registered for nextiva and below is my configuration :

When i have a discussion with their team they said that SIP is getting registered but call request is not hitting their host

Below is my configurations

sip.conf

[general]
context=default
allowoverlap=no
bindport=5060
disallow=all
allow=ulaw
bindaddr=0.0.0.0
srvlookup=yes
register => 86XXXXXX:414XXXXX@208.73.146.95/8659XXXXX

[1000]
type=friend
host=dynamic
secret=1000
context=phones

[nextiva]

disallow=all
allow=ulaw
username=8659XXXXX
fromuser=9282XXXXX
type=friend
secret=4143XXXXX
qualify=no
maxexpirey=3600
host=208.73.146.95
fromdomain=208.73.146.95
insecure=invite
dtmfmode=rfc2833
defaultexpirey=60
nat=yes
canreinvite=no
context=from-trunk

extensions.conf

[outgoing]
exten=>_x.,1,NoCDR()
same=>n,Set(TIMEOUT(absolute)=10)
same=>n,dial(SIP/nextiva/+${EXTEN},6)

When i dial the asterisk dial status : CHANUNAVAIL

Please help

2

This is an Asterisk Support, not an AsteriskNOW question.

Turn up the logging until you get a diagnostic saying why the channel is unavailable.

Whilst this contains the usual sorts of errors from using boiler plate configurations, (canreinvite, and friend), none of them should prevent outgoing calls.

Note that CHANUNAVAIL normally means that Asterisk didn’t even try to send the INVITE.