TIPS Wazo trunk config for Unlimitel

Ramblin

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@wardmundy

Your tutorials on configuring trunks and inbound/outbound routes have been very helpful

I have a vitelity trunk setup and can dial out and in with 2-way voice.

I see in your post
http://nerdvittles.com/?p=20675

that you have a number of tutorials for other trunk providers.

I have about 6 trunks from Unlimitel
(a good Canadian provider - at least they were before the Primus takeover)
and they provide a great configurator for asterisk and FreePBX at
http://www.unlimitel.ca/temp/support/voip_support/voip_configuration_samples.html
http://www.unlimitel.ca/temp/support/voip_support/voip_configuration_samples.html
I have used that to help configure my PBXiaf and it works fine.

I cannot make the translation to Wazo.

Any chance you could help with translating the FreePBX configuration shown in the configurator to what the inputs are for Wazo? In the configurator, I entered:
  • the Unlimitel server (pick any one, I can work with that as a sample)
  • DID (anything will do; I can translate - I used 5195551212 to test it out)
  • Password: MySecret
  • CallerID NAME: WardMundy
  • Server type: Choose Asterisk or TrixboxCE (Works for PBXiaf as well) for FreePBX interface
It gave this for the Trunk Out settings
Code:
disallow=all
allow=ulaw
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=sip06.unlimitel.ca
username=5195551212
secret=MySecret
insecure=very
progressinband=no
relaxdtmf=yes
rfc2833compensate=yes
type=peer

and this for the trunk In settings
Code:
disallow=all
allow=ulaw
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=sip06.unlimitel.ca
username=5195551212
secret=MySecret
insecure=very
progressinband=no
relaxdtmf=yes
rfc2833compensate=yes
type=peer

Unlike Vitelity, you enter this all in one configuration with FreePBX, but I am not sure if that is also true for Wazo.
 

wardmundy

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Have you tried following one of the other tutorials. It's pretty much trial and error. If one approach doesn't work, try another one. Let us know what works.
 

Ramblin

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I have managed to get the wazo configuration for Unlimitel to accept a call in - with 2-way voice.
I cannot yet get it to work for a call out.

I do NOT know if I have the inbound settings as they should be or if some of them are redundant / not a good idea. I duplicated some of the settings you had in the other tutorials. It did not look like it was working, I think because my IP was temporarily shutoff from the Unlimitel server for too many failed connection attempts. But when I came back from a walk with the dog, there it was, registered and working for inbound calls, (but not outbound).

Any help in both validating/correcting the inbound settings and suggesting some outbound settings would be appreciated.

So, here is what I have for the inbound settings (actual phone number and password fake):

in-General.png


in-Register.png


in-Signalling.png


in-Advanced.png
 

Ramblin

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@wardmundy

Per your question, yes, I have tried many permutations of the other tutorials to get the inbound calls working, but no luck with the outbound.
 

wardmundy

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What does the Asterisk CLI show when you attempt an outbound call? Your trunk is obviously registered if incoming calls are working so that's not the problem.
 

Ramblin

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Well @wardmundy ,

I got it working.

Part of the issue was that, after my numerous failed attempts, I was temporarily blacklisted by the Unlimitel server - I must have looked like a hacker to them. So even when I did find a configuration that should have worked, it did not, since I was blacklisted. I kept trying new permutations, which extended the blacklist, ... . Left it, came back and voila.

I posted the full configuration setup - with credit to your tutorials - at
https://projects.wazo.community/boards/1/topics/9274?r=9296

Thanks again sir.

This is looking like a very nice system!
 

wardmundy

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@Ramblin: Could you take screenshots of what you ended up with to make Unlimitel work when you have a chance. Then I'll add a new tutorial for posterity. Thanks.
 

Ramblin

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@wardmundy

Apologies for the delayed response. I am not getting emails when threads I am watching get new posts. I have checked my settings and everything appears to be set to get notifications but I don't get them.

I happened to come back in today to check and saw you had asked for the configuration pictures to be uploaded.

Any ideas on how I get that working again?

====================================================================

So, here are the updated settings that are working:

====================================================================

I do not know if all the settings I am using are required or correct, but it works. If anyone sees something I have done that is not right, PLEASE let me know. This was a "hunt and peck" operation so I am open to making it better.

A few notes up front:

1) If the Wazo server is NATed (mine is) behind a router and firewall, make sure to forward the following posts to the wazo server
Code:
5060-5069
10000-20000

This is NOT needed for inbound calls since the registration is accomplished by the Wazo server reaching out to the trunk provider server
This IS needed for outbound calls via Unlimitel to work, since they are done without the registration and (I am guessing) some packets return on another set of ports
(This is NOT needed with a Vitelity trunk, since both Outbound and Inbound are configured separately)

2) The choices in
Code:
Services -> IPBX -> Trunk management -> SIP Protocol

do NOT reflect the full suite of configuration options so you must also go to
Code:
Services -> IPBX -> General settings -> SIP Protocol -> Network (tab)

and set (replace my.actual.fixed.ip with your IP_Address)
Code:
External IP address: my.actual.fixed.ip

and then go to
Code:
Services -> IPBX -> General settings -> SIP Protocol -> Signalling (tab)

and set
Code:
Relax DTMF: Yes
Compensating for RFC 2833 DTMF from another IP PBX: Yes


3) While messing around with settings, if you try things that don't work, and keep trying, you may end up being (hopefully temporarily) blacklisted by your trunk provider's server since your repeated registration failures look like a hack attempt. This did happen to me and I had to wait for an hour to try new things. Unfortunately, it looked to me like nothing was working (which it was not because I was blacklisted) so in fact, even though I had found the correct configuration, it looked like it was not, so I kept plugging away with numerous permutations and extended the blacklist time with all my failures. A word to the wise ... (or in my case the blind and dumb)

4) Unlimitel has a very old Asterisk server (v 1.6.2.2) and their configurator actually gives configuration settings for an even older version (1.4). One specific example of why this might have been an issue was the deprecated setting of canreinvite which was renamed to directmedia (in v1.6) - thank you Pascal Cadotte-Michaud for that heads up. The newer name (directmedia) is used by Wazo. For anyone connecting to an old system, this is something to be aware of.

Now you can go to
Code:
Services -> IPBX -> Trunk management -> SIP Protocol

and set up the Unlimitel Trunk
(fake user (5195551212) and password/secret (MySecret) used)
(the server IP_Address can vary, depending on which server Unlimitel uses for your trunk; mine was sip06.unlimitel.ca)

in-General.png


in-Register.png


in-Signalling.png


in-Advanced.png



You can then check the status of the registration from the CLI by entering
Code:
asterisk -rx "sip show registry"

and you can check to see what channels are being used (during a test dial) by entering
Code:
asterisk -rx "sip show channels"
 

Ramblin

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An Update:

The above settings will work to connect a trunk to Unlimitel.

However, there are a couple of things I have experimented with since that I now use and wanted to update the recommendations.

The above will identify one trunk that is used for both incoming and outgoing calls. Although it works, I prefer to have one trunk to monitor for incoming and one trunk to monitor for outgoing calls. It makes trouble-shooting and system monitoring better.

Also, the recommendation to forward ports 5060-5069 (SIP Registration) and 10000-20000 (SIP Audio) is NOT needed if you are not allowing direct access to your server from extensions at satellite locations but instead only allow access to your server from the trunk providers. In my case, I do use satellite locations, so I do forward these ports, but I wanted to clarify. By NOT setting the ports to forward, the system is natively more secure. If you do forward the ports, make sure to set up your firewall to only allow a whitelisted set of addresses, and, if your satellite IP addresses change, use Dynamic IP Address updating for your firewall, like that seen in the IncrediblePBX TravelinMan.

So, if you do want the Unlimitel trunk to show two connections, one for inbound and one for outbound calls:
  • Go to Services -> IPBX -> Trunk management -> SIP Protocol
  • Click on the Unlimitel trunk setup using the above instructions
  • Change the name of the existing trunk, configured exactly like above, to in-5195551212 (change to match your specifics)
  • Create a second trunk, with exactly the same settings as above, with the following exceptions:
In Services -> IPBX -> Trunk management -> SIP Protocol -> General (tab) -> Name: out-5195551212 (change to match your specifics)
In Services -> IPBX -> Trunk management -> SIP Protocol -> General (tab) -> Context: Default (default)
In Services -> IPBX -> Trunk management -> SIP Protocol -> Register (tab) -> Register: Uncheck (you can leave the rest of the entries in this tab blank)​


Now, if you monitor your trunk usage (using FOP2 or asterisk CLI commands, you will see two trunks:
  • one for inbound traffic
  • one for outbound traffic

and I find this helps to better monitor and manage your system
 

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