[asterisk-users] Testing a sip call through Asterisk?

[Nathan Clemons]

I'm looking to find a test tool that will register with our Asterisk
(Trixbox) server here at work and place an outgoing call via our main SIP
trunk (BroadVoice) to confirm that things are working. I've looked around
but I can't seem to find any tools that will do what I'm looking for.

I can't just monitor the status of the trunk inside Asterisk, as this is the
normal status:

asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port
Status
BroadVoice/425256XXXX      147.135.32.221       N      5060
Unmonitored
...
37 sip peers [Monitored: 5 online, 31 offline Unmonitored: 1 online, 0
offline]
asterisk*CLI>

Alternatively, any suggestions as to how I can change the trunk
configuration so that it is monitored would be appreciated. The peer config
is set as:

allow=ulaw
disallow=all
canreinvite=no
context=from-trunk
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=425256XXXX
host=sip.broadvoice.com
insecure=very
nat=yes
secret=XXXXXXXXXX
type=peer
username=425256XXXX


Any assistance would be appreciated. I'd rather know when things fail via an
automated system rather than learning it's down from the users.

-- Nathan Clemons
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