Then there is something wrong wrong wrong in your setup. Do you see NO templates? If not, you need to acknowledge if you have
EnabledNameUse built-in SIP Trunk SBCUse provider template 4.2 was almost no different. If you have trunking role enabled, it shouldshow an option (4.2 was a little different) in that you had to choose the sipXbridge-1 selection from the dropdown. Do us all a favor and look at creating a siptrunk/gateway and seeing what options you have there. On Mon, Mar 26, 2012 at 6:10 PM, Stiles Watson <[email protected]>wrote: > It is not there. I've tried Devices>Gateways>Add new gateway... a dozen > times. I've restarted all the services, I've rebooted the server, even > reinstalled... It is not there. I'm using Firefox 11 on Ubuntu 11.10. I'm > also using Chromium (not supported) on the same OS. I've tried both FireFox > and IE in Windows XP Pro, it is not there. > > To comment on Tony's reply, I have a Sonicwall NSA 240 firewall. I have > SIP transformations disabled. I have Consistent NAT enabled. I've opened > ports 5080 UDP, 5060 UDP & TCP (for remote phones) and 30000-31000 UDP for > RTP. I've also created the NAT policies to direct WAN traffic on these > ports to the sipx server. All trafic going out to the WAN is allowed. I > have connection limiting on 5060 to prevent a SIP DoS. > > I have not downloaded a new iso lately. I can try that next. Should I > stick with 4.2 or go to 4.4? I'm using Polycom phones. > > Stiles > > > On 03/26/2012 05:12 PM, Todd Hodgen wrote: > > 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]<[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**** > > > _______________________________________________ > sipx-users mailing [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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