asterisk users - Apr 2010 - Testing a sip call through Asterisk?

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100416/e2eb700c/attachment.htm

On 04/16/2010 03:39 PM, Nathan Clemons wrote:
> 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 <http://sip.broadvoice.com>
> fromuser=425256XXXX
> host=sip.broadvoice.com <http://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
I believe that adding qualify=<enter your value in seconds here> to your 
trunk configuration is what you are looking for for the monitoring 
state.  This will send SIP OPTIONS packets to the trunk periodically.  
See "qualify" in the sip.conf samples or documentation.

 From there you can use a monitoring solution to monitor the state of 
the trunk.  Alternatively you can use a OSS tool called SIPp to test SIP 
devices.  See *http://sipp*.sourceforge.net for more information.  This 
is an indispensable tool for SIP and Asterisk troubleshooting.

I hope this helps.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100416/9f549dab/attachment.htm

On Fri, 16 Apr 2010, Nathan Clemons wrote:
> 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:
>
[snip]

just add "qualify=yes" to your context and it will monitor the RT
latency.

j