Choose sipxbridge then hit apply when creating the sip trunk. On Mar 26, 2012 6:17 PM, "Tony Graziano" <[email protected]> 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 <[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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