Walking through Tony's voip.ms how-to. All my notes are delimited by
---> <---and are in /_italics and underlined_/.
**
*Dealing with Step 3, online with voip.ms*
At the voip.ms portal:
Main Menu > Account Settings (for a main account, not subaccounts)
>Account Restrictions
Adjust the call timer restrictions here for US and International calls
as desired.
--->_/Made no changes to the defaults/_<---
1. Click GENERAL>Music on hold = No Music-Silence [APPLY]
--->/_Done_/<---
2. Click INBOUND SETTINGS > Protocol = SIP--->/_Done_/<---, Device
Type = IP PBX Server, Asterisk or Softswitch--->/_Done_/<---
(otherwise ALL your DID calls use the account number in the invite).
[APPLY]
3. Click DEFAULT DID ROUTING>Choose the default city your calls
should go to when setting up new numbers--->/_Done_/<--- and what
account/subaccount should be used by default for new
numbers--->/_Done_/<---. [APPLY]
4. Click ADVANCED>NAT = No--->/_Done_/<---, DTMF Mode = AUTO (or
RFC2833, either is essentially the same with sipx, since it only
uses RFC2833/sip)--->/_Done, chose AUTO_/<---, Allowed Codecs =
G.711 (uncheck the others)--->/_Done_/<---[APPLY]
After you purchase a DID number, ensure it is pointed to the city where
you have a registration and the account associated with that
registration (We'll use Atlanta in this example).
Account 123456 is my main account with voip.ms. So when I create or edit
DID 4345551234 I make sure it points to SIP/IAX account [123456] and set
the DID Point of Presence for "Atlanta, GA". Change the dialtimeout to
300s, and [APPLY].--->/_Done, purchased DID, pointed it to my account
and presence of Atlanta, GA_/<---
*Dealing with Step 4, in sipxconfig.*
We will create the gateway, apply it, register it, confirm it at both
sides instantly, assign a DID and send and receive a call.
Create the Gateway. I'll make it easy with screenshots:
Devices>Gateways>AddNewGateway (link at top right), choose SIP Trunk
--->_/NOTE: Screen shot shows a "User provider template" drop-down, but
this drop-down does not exist on my Gateway Details>Configuration
screen!//I am using 4.2.1-018971.21.0/_ <---
enable it--->/_Done_/<---, give it a name--->/_Done_/<---, and choose
the voip.ms template from the list--->/_Does not exist_/<---, change the
"address to match the city name (i.e. atlanta.voip.ms)--->/_Done_/<---,
CLICK APPLY.--->/_Done_/<---
Now set the dial plan up in sipxecs for outbound calls....
--->
I did not do this. I changed the digitmap under Devices>Phone
Groups>group_name>Polycom SoundPoint IP 335>Line>Dial Plan to make sure
the number was dialed correctly.
Digitmap:
[2-9]11|0T|RR9R011xxx.T|9011xxx.T|RR91R[2-9]xxxxxxxxx|RR9R1[2-9]xxxxxxxxx|91[2-9]xxxxxxxxx|RR91919R[2-9]xxxxxx|91919[2-9]xxxxxx|*xx|[8]xxx|[1]xx.T
<---
Now finish the gateway config for the ITSP account.
--->Image removed<---
There are three fields here. username/authentication username. These are
the same values, which is the account/subaccount number you have with
voip.ms--->_/Done/_<---. The password is the sip password (not the
portal password) in your voip.ms portal for the
account/subaccount--->_/Done/_<---.[APPLY]
You will be asked to restart several services, you should do so and then
wait 15 seconds or so and check to see if it is registered--->_/Done/_<---.
Go to Diagnostics>SIP Trunk SBC Statistics
--->_/Image removed/_<---
If you did this correctly the account will show
registered--->_/Done/_<---. NOW, go to voip.ms and see if they concur
and have the proper IP:port listed.
At the voip.ms website, login, Portal home page...it should show a green
REGISTERED State --->_/Done/_<---. Hover over the dot to the right of
registered, You should see your public IP address that sipx is using
(you did this setting up the firewall porting, system>server>NAT and set
the static IP here or are using STUN to determine it)--->_/Done, using
static IP/_<---. The IP should show your port as "5080?--->_/Done/_<---.
if it does not, you should go back and address your firewall configuration.
Dialing out it simple.
Dialing in requires the DID be put in the service DID field or user
ALIAS field in the format of NPANXXYYYY (4345551234). If you used this
for an auto attendant or other service, you will need to restart
services prompted in order to apply this setting, user aliases do not
require services restart/reload--->_/Done, I added the voip.ms DID as an
Alias to the default Auot Attendant/_<---.
You should be able to set the default caller ID in the gateway (if it
needs a glocal setting, or leave blank and set the caller ID in each
user line as desired, don't leave both blank).
Congratulations, you have trunking and DID services setup without any
paperwork in 15 minutes!
--->_/Done, except for retrieving hold and canceling transfers/_<---
Stiles
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