You are missing something with your Gateway setup. If you go to Gateway, and click on the box with “add new gateway” and select SIP trunk it will open a new gateway configuration screen. 4th item down is the templates selection box………………………
From: [email protected] [mailto:[email protected]] On Behalf Of Stiles Watson Sent: Monday, March 26, 2012 2:05 PM To: Discussion list for users of sipXecs software Subject: [sipx-users] voip.ms config 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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