SIP Trunk Outbound Call Issues

General Help
1

Been a year or so and I am kind of rusty.

Got everything setup good, but no outbound audio on my Nextiva Trunk. Call connects, but no ringing or audio on either end. Inbound works fine. Another trunk with another carrier works fine so that pretty much rules out NAT problem.

Anyone see something little that I am missing?

outbound callerid: USERNAME
Force-Trunk CID

PEER DETAILS

username=USERNAME
fromuser=USERNAME
type=friend
secret=PASSWORD
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

Incoming Settings
user context: USERNAME

USER DETAILS:

secret=445275560
type=user
context=from-trunk

Registration key: USERNAME:[email protected]/USERNAME

2

whoops also have

disallow=all
allow=ulaw

at the beginning of PEER DETAILS

3

Incase someone else needs help with this in future here is update. I tried different things in Peer Details and looks like it is working fine with:

disallow=all
allow=ulaw
username=XXXXXXX
fromuser=XXXXXXX
type=friend
secret=XXXXXX
qualify=no
maxexpirey=3600
host=208.73.146.95
fromdomain=208.73.146.95
insecure=invite
dtmfmode=auto
session-timers=refuse
defaultexpirey=60
nat=no
canreinvite=no
context=from-trunk

4

Ok cancel that it only worked for a few times. Now im back to

disallow=all
allow=gsm&ulaw&alaw
username=USERNAME
fromuser=USERNAME
type=friend
secret=SECRETKEY
qualify=no
maxexpirey=3600
host=208.73.146.95
fromdomain=208.73.146.95
insecure=invite
dtmfmode=auto
session-timers=refuse
defaultexpirey=60
nat=no
canreinvite=no
context=from-trunk

outbound call audio still not working. Ahhh!

5

Are the phones and the server in the same network?

You don’t need the insecure statement, you have a username and password. The two are mutually exclusive.

Also alaw is Europe ulaw is North America, specifying both is confusing.

6

Yep, phones and server are same network. Running Virtual Box.

7

Ok, sounds like a router problem to me. You may need to open RTP ports to your carrier. Default is 10000-2000. You can edit in /etc/asterisk/rtp.conf You need 2 per concurrent call leg and 20% headroom.