Correct. On Mar 27, 2012 12:05 PM, "Stiles Watson" <wat...@datatek-net.com> wrote:
> This is where one swallows one's pride.... The way I was entering data > caused the drop-down to not be displayed. > > To keep this short: > > 1. When you first select Add new gateway>Sip Trunk, the template drop > down is not visible. I was not aware this was the case until yesterday. I > just thought it was not there. > 2. The template drop-down is only displayed after you enter a name for > the gateway and then select the default SBC. > 3. If you ever click the Apply button before both the name and SBC > are entered, the drop down is never displayed. > > This is why I never saw the template drop-down. > > Now, having said all of that, I deleted my existing voip.ms gateway and > created a new one using the template drop-down. However, this did not fix > my problem and everything is as it was before. I still can not retrieve a > call from hold or cancel a transfer. I have verified in my voip.msaccount > that it is registered with the public IP and port 5080. > > So it looks like we are back to a firewall problem, correct? > > Stiles > > On 03/26/2012 06:52 PM, Tony Graziano wrote: > > Choose sipxbridge then hit apply when creating the sip trunk. > On Mar 26, 2012 6:17 PM, "Tony Graziano" <tgrazi...@myitdepartment.net> > wrote: > >> Then there is something wrong wrong wrong in your setup. >> >> Do you see NO templates? If not, you need to acknowledge if you have >> >> Enabled >> Name >> Use built-in SIP Trunk SBC >> Use 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 <wat...@datatek-net.com>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:* sipx-users-boun...@list.sipfoundry.org [ >>> mailto:sipx-users-boun...@list.sipfoundry.org<sipx-users-boun...@list.sipfoundry.org>] >>> *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 listsipx-us...@list.sipfoundry.org >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> >>> >>> _______________________________________________ >>> sipx-users mailing list >>> sipx-users@list.sipfoundry.org >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> -- >> ~~~~~~~~~~~~~~~~~~ >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgrazi...@voice.myitdepartment.net >> 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: helpdesk@voice.myitdepartment.**net<helpd...@voice.myitdepartment.net> > > Helpdesk Customers: > http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> > Blog: http://blog.myitdepartment.net > > > _______________________________________________ > sipx-users mailing listsipx-us...@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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