Frequently Asked Questions

Go to the link https://ccm1.downetworks.com/ccmuser.  There you will need to enter in a username and password.

Once you log in, you will see a list of options to choose from.  Select the first option, “Forward all calls to a different number.”  There you will see a list of your numbers and the locations they are currently pointing to.  To change the routing on them, change the number to where you want the calls to terminate in the “this number” field.  Once you have made your changes click the update button.

When re-routing calls please follow this dialplan:

• Routing to numbers in the US, Canada and Caribbean:  1+ 10 digit number (including the area code)
• Routing to numbers outside of these areas:  011+ country code + city code + number

SIP, H323, MGCP, SCCP, and IAX2

• H323 (G729, G729b, G711ulaw and G711alaw)
• SIP  (G729a, GSM, G711ulaw and G711alaw)
• MGCP (G729, G711ulaw and G711alaw)
• SCCP (G729, G711ulaw and G711alaw)
• IAX2 (G729a, GSM, G711ulaw and G711alaw)

The best way to implement VoIP security is to run the traffic through an encrypted VPN tunnel.  This way the terminating voice devices do not bear the burden of encrypting and decrypting the voice, which results in use of device resources.  We do orginate and terminate VPN to our network.  As a secondary option, we can also run a GRE tunnel.

We prefer to send a real time ANI for all outbound calls.  This number can either be toll-free number or DID, which we provide, or a local DID or DDI provided to us.  We cannot currently support caller ID names as this is controlled by the LEC of our telephony provider.

Once your licenses have been purchased, they can be installed by clicking on the link in the original email sent from the reseller, which takes you to the registration page.  You then need to put in your registration code found on the Switchvox software jewel case and then enter all subscription codes.  Once you click the button to "Activate" them, you will then go into your Switchvox admin page and click on "Machine Admin => Updates."  There it will refresh all available updates.  Then click on "Apply this Update" next to the subscription update.  Make sure you make any updates during off-peak hours because it will automatically restart the software, which both drops all current calls as well as logs all logged-on agents.

In order to log an agent into their respective queues in QueueMetrics, they must do the following:

1. Open a web browser and point it to address http://<FQDN of QueueMetrics server>:8080/queuemetrics.  *The Fully-Qualified Domain Name of the QueueMetrics server should be provided to the customer upon service set up.
2. You will then be sent to a "User Login" page.  In the field for "Login" you should type "agent/<agent ID>" (Example: agent/0080).  Then enter the password, which should be the same as the agent ID.  After clicking the "Login" button you will be taken to the agent page.
3. Click on the link that says "Show inbound calls for agent Agent/<agent ID>.  On this page you will see the current status of the agent (logged on, logged off or paused).  To log an agent on simply click the button labeled "Log on."  It will then prompt you to enter the extension which should be the same as the agent ID or agent code, then click the "Run" button.  It should then change the agent status to "Agent <agent ID> is currently logged on."  To log an agent off click on the "Log Off" button and then "Run."
4. Pause should be used when agents wish to not receive calls for whatever reason during their shift.  When an agent  clicks on the "Pause" button it will give them a list of reason codes from which to select the reason for going into pause (Example: Lunch, Hourly Break, etc).  While an agent is in "Pause" they will still be logged into queue but will not receive any calls.  In order for the receive calls again they must click on the "Unpause" button.

If you add a .WAV file to Asterisk and it won't play when you use the "Playback" or "Background" application you will need to "Sox" the file.  The command syntax goes something like this:

• sox -V <filename.wav> -r 8000 -ch1 -w <newfilename.wav>
• V = Verbosity shows the different phases of the conversion
• r = This switch converts the file hz. 8000 hz is suggested for all recordings on Asterisk as well as recordings on other platforms like IPCC
• c1 = Converts the file to mono
• w = Changes the file to a 16 bit file.

     http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk

When attempting to set up G729 pass-through on an Asterisk server with no G729 licenses installed you must make sure that G729 is the only codec running on both channel (endpoint) configurations on the Asterisk.  If not you will have problems such as calls not completing and one-way voice.  Once you disable all other codecs and just have G729 configured for each Asterisk channel then your G729 pass-through calls should work properly.

There are two ways we can send DTMF tones to an enpoint device out-of-band (SIP INFO or RFC2833) or inband.  Out-of-band is sent in the VoIP protocol signal or message.  Inband is when the tones are sent in the RTP (voice stream).  It is suggested that the DTMF be sent out-of-band as sending it inband requires using an "uncompressed" codec such as G711 ulaw or alaw.