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. 4^th 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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