bfacer,
The manual you found, like most of them created by Mitel and other Vendors, follow an odd path in my opinion. They typically start in the middle and work their way out. I for one find it easier to start on the outside and work my way in to ensure that everything is set when it can be set instead of going back and forth.
Typically I start on the IP Call Configuration and create a new one just for the SIP Trunk Group that will be created later on. This ensures that if I need to make changes to either the IP phones or SIP Trunks configurations later that I don't change anything for the other devices; segregation is the key when testing.
Go to System\IP-Related Information\Call Configurations\
Right-Click and select [Add To Call Configurations List...]
Take the next available Starting ID, typically 3 on a new database, and click [OK].
Single Left-Click on the Description and Type in the name of the SIP Provider for whom the SIP Trunks will be created for.
Next double left-click on the newly created Call Configuration.
System\IP-Related Information\Call Configurations\3\
Inside here we will need to get some information from your SIP provider and/or the MBG.
Now, the main items within a Call Configuration are as follows.
1. Audio Frames/IP Packet or what is commonly referred to as RTP Framesize or the Packetization Interval or ptime for short. These will be stated in terms of milliseconds from the SIP Provider and are typically 20 ms or 30 ms. The default value on a newly created Call Configuration is set to 3 which means 30 ms. So if they tell you they need a max ptime of 20 ms you need to set this to 2.
NOTE: The Minimum Playback Time is usually adjusted to reduce jitter when playing the audio and you will notice in Call Configuration 1 it is set to 20 so there is less latency in the communication with a higher chance of jitter due to packet loss due to network latency. Now if you look at Call Configuration 2 and 3 it is set to 80 ms which means there is more latency in the audio, but less likely that there will be jitter due to packet loss from network latency. Although Audio Frames/IP Packet and Minimum Playback Time do two different things they are sort of tied to together when looked upon as a whole. I have no suggestions for what to put here as it is purely up to you what works best for the customer and their network.
2. DTMF Encoding Setting. By default this is set to G.711 Mu-Law in US databases. Typically this is determined by the SIP Provider and for the most part they will suggest RFC 2833.
3. Speech Encoding Setting. By default this is set to G.711 Mu-Law in US databases. This is again determined by the SIP Provider and for the most part they will suggest G.711 Mu-Law.
4. Fax Detection Sensitivity. Unless you know for sure there will be no Faxing coming in via the SIP Provider then set this to at least 1.
Alright on to the next step creating the SIP Peer Trunk Group.
Go to System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\
Right-Click and select [Create SIP Trunk Group].
Use whatever number scheme fits your programming and click [OK]. On a default database this will be 92002.
Once it is created then single left-click on Description and Username and give them appropriate names. Typically under Description I will put the SIP Providers name and under Username I will put [No Call ID]. This goes back to the typical calls coming in with no Caller ID information so it would show what was in the Username; do what is appropriate for your site.
Next double left-click on the Extension to go further into programming.
System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\
After that you should have two choices Configuration and Trunk Group Configuration. Let's go into Configuration first.
System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\Configuration\
Under this is where the magic happens; sort of. Since we have already defined part of how it will work in the Call Configuration then make sure that you change the Call Configuration to the ID of the one created earlier for the SIP Provider; in the case of this example it would be set to 3.
Typically there are a few items that you will need to know before going forward with the programming.
1. The FQDN of the SIP Providers SIP service; never use the IP Address if you can get away with it. This does require that the DNS Server IP Addresses under IP Settings go to DNS servers that can resolve the FQDN provided.
2. Does the SIP Provider require the 5000 to use ITU-T E.164 when formatting the called number for out-bound calls? The 5000 supports ITU-T E.164 on in-bound calls regardless of this flag.
3. What DTMF Decoding Payload do they use? This is very important in determining as it determines how DTMF tones are being received and generated. Typically 96 and 101 are the values used, but could be anything in the range from 96 to 127.
System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\Configuration\Registrar
4. Do they require Registration? What is the registrar FQDN? Again try not to use an IP Address if you can get away with it. This does require that the DNS Server IP Addresses under IP Settings go to DNS servers that can resolve the FQDN provided.
System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\Configuration\Authentication
5. Do they require authentication on Out-bound calls? You typically do not require In-bound Authentication on SIP trunks, but you can request that the SIP Provider or the MBG do so. If you do require it then make sure you have a unique username and password.
System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\Configuration\Alternate IP/FQDN List
6. Are there other IP Addresses or FQDNs that calls would come from/sent to the Mitel 5000?
System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\Configuration\Route Sets
7. Is there another route we should take when going to the FQDN or IP Address of the SIP Provider. In this case the answer for you would be YES it goes through the MBG.
At this point you are about half way through the programming of the 5000, but the rest is just the basics of call routing. You may have noticed that the Maximum Number of Calls is still set to 0 under Configuration, but that will be taken care of when the Trunk Group Configuration is completed.
Go to System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\Trunk Group Configuration\
Hopefully this page looks familiar as it is similar to the normal CO Trunk Groups setup page. If it doesn't then most likely you haven't programmed too many 5000 systems, but not to fear you can do a star and compare to the customers original PRI CO Trunk Group and get all if not most of your information.
The biggest things to look at are Day Ring-In Type and Night Ring-In type as those should typically go to Call Routing Table as their destination so you can set routes for individual DIDs or DDIs depending on what side of the pond your on. If you are going to be porting over all the DIDs/DDIs of the PRI then just use the same Call Routing Tables that the PRI Trunk Group goes to and the hard work is done.
The next thing you will want to set is the Propagate Original Caller ID to Yes. This allows you to send the Caller ID of an In-bound Call back out as though it was a number that was owned by the customer. This also requires that you set the Propagate Original Caller ID on any devices that you want to allow that feature for; this flag is the main flag for turning the feature on for the Trunk Group.
If you have any special Music On Hold then go ahead and set that up as required for the customer.
Next we go to the Trunks so we can create them. This is how the Maximum Number of Calls will go from 0 to however many trunks you can create.
System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\Trunks\
Right-Click and select [Create SIP Peer Trunk].
Select a starting extension and then change the Number of Extensions to reflect how many SIP Trunk licenses you have or want to add to this particular SIP Trunk Group and then click [OK].
Once your SIP Trunks are created then single left-click on the Label and label each one however you want. This is purely cosmetic and only holds weight if you are doing some sort of reported or troubleshooting. I still suggest doing it so you can reduce your headaches later on.
Also if you don't trust me you can go back and look at the Configuration folder and see that the Maximum Number of Calls has raised from 0 to however many trunks you created.
Another thing that I do typically for troubleshooting purposes is change the Exempt from ARS Only flag from No to Yes. This will allow you to direct dial a trunk extension ID or trunk group ID and dial direct without ARS involved.
System\Devices and Feature Codes\SIP Peers\SIP Trunk Groups\92002\Toll Restriction\Exempt from ARS Only
Speaking of ARS we will now need to go to Number Plan and make sure that the system will now use our newly created CO Trunk Group.
Go to System\Numbering Plan\Facility Groups\
Open up each of the Facility Groups and go to their Trunk Groups/Nodes folder.
System\Numbering Plan\Facility Groups\{P####}\Trunk Groups/Nodes\
Right-Click and select [Add To Trunk Groups/Nodes List...]
Select SIP Trunk Group from the Menu Box and then click the Next button.
Select the Trunk Group(s) that you want calls to go out of for this Facility Group, which is basically a Call Type, and then click Add Items followed by Finish.
If you have a need for multiple CO Trunk Groups then just make sure they are in the order you want them used in as this is a top down search list for trunks. If you need to remove a CO Trunk Group then select it by right-clicking on it and then selecting [Remove Selected Items].
If you are not going to reuse the original PRIs DIDs/DDIs then it would be best to go to Trunk-Related Information and fill in the desired routes on the tables listed under the Day and Night Ring-In Types under your SIP Trunk Group Configuration.
This little tutorial is by no means the end all be all of how to program SIP Trunk Groups on a 5000, but it should at least point you in the right direction for most of your questions.
Thanks,
TE
