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.ms account 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" <[email protected]
<mailto:[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] <mailto:[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. 4^th item down is the templates selection
box...........................
*From:*[email protected]
<mailto:[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 <http://voip.ms> config
Walking through Tony's voip.ms <http://voip.ms> how-to. All
my notes are delimited by ---> <---and are in /_italics and
underlined_/.
*Dealing with Step 3, online with voip.ms <http://voip.ms>*
At the voip.ms <http://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
<http://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 <http://voip.ms> template from the
list--->/_Does not exist_/<---, change the "address to match
the city name (i.e. atlanta.voip.ms
<http://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 <http://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
<http://voip.ms> and see if they concur and have the proper
IP:port listed.
At the voip.ms <http://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
<http://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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Telephone: 434.984.8430
sip: [email protected]
<mailto:[email protected]>
Fax: 434.465.6833
~~~~~~~~~~~~~~~~~~
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LAN/Telephony/Security and Control Systems Helpdesk:
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