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]>
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