SIP Trunking Issue / Question

[niddnet]

Hello all - my first post - I'm putting it in here, because it's specifically 5000 related, rather than more generally SIP related:

I am trying to utilise SIP Trunking on a Mitel 5000 - v6.  Licences are in place.

I am having issues with getting the system to authenticate and, as I'm working remotely, I'm not able to see exactly what strings the Mitel is sending to the provider.

a) Has anyone successfully used Twilio SIP trunking on the 5000 - and if so would you be willing to share a config?

b) Is there any way I can monitor (ideally, at packet level) the SIP conversation between the 5000 and Twilio, **without** being physically on site.

I have access to the LAN, but the site is very non-technical, so I'm not able to direct someone to assist in packet captures - it would likely make them explode.

Twilio have provided some indepth info here:  https://www.twilio.com/docs/sip-trunking  -   and also, some sample configs here:  https://www.twilio.com/docs/sip-trunking/sample-configuration

I'm just having REAL issues converting this information into Mitelese.

Hope someone is able to help out - many thanks


R.

[Tech Electronics]

Niddnet,

Alright I have never heard of Twilio, but it doesn't seem that it should be all that hard to make it work; at least the basics. As I read over their current guides to get others systems up and working I noticed they have a few common settings so we will start with those and see if we can't get you connected.

First off start by making yourself a new Call Configuration. System > IP-Related Information > Call Configurations: Right-Click in the area on the right and create a new Call Configuration; name this Twilio PSTN or whatever makes sense to you.

Within that Call Configuration you will need to match what they are looking for.

Transmit DTMF Level: United Kingdom <- Really all but Japan uses the same DTMF Level, but it doesn't hurt to change it.
DTMF Encoding Setting: RFC 2833
Speech Encoding Setting: G.711 Mu-Law <- Not sure why they use this in Europe but all the examples have it set that way. If you are having audio issues then change this to A-Law which is what is used in Europe.

The rest of it doesn't seem as though it is a concern to them; at least according to the other configurations.

Now, go to your SIP Peer Configuration. System > Devices and Feature Codes > SIP Peers > SIP Trunk Groups > {your group} > Configuration: Only a few things to change on the front page from what I can see on the other guides.

Fully Qualified Domain Name: your_business.pstn.twilio.com
Call Configuration: x <- Use the number of the configuration your created earlier
Use ITU-T E.164 Phone Number: Yes
Static Binding: Yes

Now for Registration: It is saying that Registration is not needed. If you seem to be having issues then just Enable Registration and put in the FQDN above: your_business.pstn.twilio.com

Next go to Authentication: They don't seem to care about In-Bound Authentication, but if it seems it won't work on In-Bound calls then use the same credentials for Out-Bound

Out-Bound Username: Username <- Your Twilio IP Credentials
Out-Bound Password: Password <- Your Twilio IP Credentials

On to NAT Settings: This kind of depends on how you have it setup in your network. The two options are pretty self-explanatory so I won't go over them here.

After that we have Alternate IP/FQDN List: They do provide a list of their IP Addresses that if you want to put them in here you can, but it shouldn't be necessary since we are using the FQDN they provided you.

Last but not least is Route Sets: This is not used in this setup from what I can see as they provide no Proxies that you must transverse to get to them.

Now you will need to go to your Trunk Group Configuration and set that up.

System > Devices and Feature Codes > SIP Peers > SIP Trunk Groups > {your group} > Trunk Group Configuraiton:

First things first you will need to go to Trunks and create as many trunks as you have provided by Twilio; don't worry about the extension numbers they provide you.

Next go to the Day and Night Ring-In Type: This tells the system where you want calls to go to. Since I don't know the setup of what types of trunking you are getting I can't really say what you will need here. Most of the time though it will be Single or Call Routing Table. A Single destination means all calls will go directly there regardless of whatever information is sent to us. The Call Routing Table is used if they are sending DID information that you want to route based on the number called; most of the time as there are reasons to use this for other scenarios.

You may also want to fill in your Calling Party Name and Number just in case they are looking for something here.

Hope that gets you going in the right direction.

Thanks,

TE

[niddnet]

Hi TE

Thanks for a nicely concise guide - I *think* I'd covered most of your points in my previous attempts however.  I'll knock it down to zero again, and start again from scratch.

Over the weekend, I ran up the SIP trunks on a test-bed of 3CX to see what was going on.  The one thing I noticed is that ALL numbers in all (both?!) directions are sent using the full international format.  When I tried to send 01234567890 the call failed - when I tried 00441234567890, again, it failed.  The only way it works is when Twilio receives +441234567890 as the string.  Can I force the 5000 to do this?

Also - is there any way I can achieve the SIP traces "officially"?  I'm more than handy with tcpdump / Wireshark, and following the conversation at packet level, but I'd have thought there should be some way to enable some verbose SIP logging somewhere?

If you're into SIP at all, Twilio is pretty nice.  Definitely worth a play with, for the money it costs.

Really appreciate your help thus far - hopefully I'll have some time to get back to it this week.

Today I'm standing atop a hill, to see how many internets one of our WISP partners can beam across to us - for another of our existing Mitel clients.

Many thanks


Ross

[Tech Electronics]

Niddnet,

From my understanding the ITU-T E.164 is what would be used for the International number portion as it is the standard for that. As for what the 5000 actually sends when it is setup for UK or AU I am at a loss since I work in the US and don't really deal with that portion of a project as we expect our counterparts in those areas to know what is needed.

Now for the built-in SIP logging that you are asking for. We do have it, but it is not very verbose and may not provide you with the information you are looking for. When I get to my lab this morning I will find the paths that need to be turned on, but it will require you to turn on the Online Monitor portion under View to see the options.

Thanks,

TE

[Tech Electronics]

niddnet,

Alright, so go to View and turn on Online Monitor.
Next, go to System > Devices and Feature Codes > SIP Peers > General Configuraiton: make the following changes.

1. SIP Message Output Format: Full
2. SIP Log Level: Debug

Now go to System > Maintenance > Message Print > Print Expanded Message Print: YES

This should allow you to see the SIP Information in the Message Print.

Thanks,

TE

[michaelFF]

For what it's worth, I'm in the same boat as niddnet was. The Mitel 5000 (MiVoice 250) has worked great for us, but we're looking to switch from a Comcast PRI trunk to a Twilio SIP trunk. We haven't gotten it to work after several hours of trying.

I was hoping when I first found this thread that there had been some resolution, whether good or bad. I'm going to assume niddnet never got it working.

[Tech Electronics]

michaelFF,

What exactly is happening when you try to set it up? There shouldn't be any reason you can't make it work with the MiVO-250 unless they are using DNS SRV and even then you can make it work if you know the FQDNs associated with it or you have a MBG at version 10.0+.

Thanks,

TE