BCM & VOIP.ms 1

[snd1234]

I recently got my BCM450r6 SIP trunks working with VOIP.ms. I wanted to share the solution here.

First, I'm not going to reinvent the wheel. I follow this guide almost exactly:

http://wiki.voip.ms/article/NortelBCM

This was also helpful as a resource but the wiki guide is custom for VOIP.ms:

https://downloads.avaya.com/css/P8/documents/100155746

The only problem with the wiki is that it doesn't address destination codes or routing which is actually not very intuitive and also really important if you want to dial out. So I'm going to pick up there.

Basically, once you have your VOIP.ms account registered in the BCM and can receive incoming calls you need to create a route and destination code. A destintation code is just the number your users press on their handset to get them an outside line. Like how in most places you dial 9 to access the PRI or the Pool of analog lines.

In my scenario my VOIP trunks are BlocF. This is set in LINES > ACTIVE VOIP LINES

So I went into TELEPHONY > DIALING PLAN > ROUTING

In the "routes" tab I created a route 009. (The specific route number doesn't matter.) I set "use pool" field to BlocF (use whatever Bloc your VOIP trunks are in)
In the "destintation codes" tab I created a destination code "88" and put route "009" in the normal route field.
***You can & should use your own code but my users will dial 88 to access the VOIP trunk.
In the "Second Dial Tone" tab I put "88" because I want my users to hear a dial tone after pressing 88.

After doing all this I kept getting a fast busy. I simply didn't understand what the problem was. I worked on it for hours and eventually I figured out that you ALSO have to set destination codes in the SIP TRUNKS configuration window as well. This was not at all clear to me from the Avaya guide.

So I went back into RESOURCES > IP TRUNKS > SIP TRUNKS and drilled down into the "Public" Tab and then the "Routing" table subtab. I pressed add

Name: voipms
Destintation digits: 0 1 2 3 4 5 6 7 8 9 (spaces after each digit except the last one)
ITSP Account: (name of the SIP profile I made early on)

Then everything started working. As best I can figure, pressing 88 on a handset only sends your call out the VOIP trunks. The system still needs to know what to do with them. By putting 0-9 in the SIP routing it tells the system to to

Pics attached.

Hope that makes sense.

 

[snd1234]

Just to clarify some things.

The destination code (eg: 88) in TELEPHONY > DIALING PLAN > ROUTING just putr your onto your VOIP trunks. The system still needs to know which SIP or H323 trunk to use. That's what the destination codes in RESOURCES > IP TRUNKS > SIP TRUNKS are for.

I probably overkilled it by using 123456789 but I'm only using one provider so that forces all the calls out that SIP trunk.

 

[snd1234]

So the 1234567890 configuration in the IP TRUNKS routing was way overkill.

You just need to give it one number and then go into the IP TRUNKS > PUBLIC > ACCOUNTS > ADVANCED section

There is a setting down the page called "outbound called characters to absorb" which is where you can get that routing code absorbed.

So, to review, make your main destination code in the DIAL PLAN section that routes you to the VOIP trunks. Set that code to absorb as many characters as necessary. Then create the secondary destination in the IP TRUNKS section. Set that code to absorb what you need it to.

For example, say you want your users to dial 82 to route calls out VOIP.ms. So if a user was going to call 1800

http://WWWDELL

on the VOIP.ms trunk they would dial 821800

http://WWWDELL.

The DIAL PLAN destination code pointing to the VOIP trunks route should be 82 with 1 character being absorbed. Then in the IP TRUNKS routing table you should set the VOIP.MS account to be 2. Then go into the account>advanced section of the VOIP.MS account and set it to absorb 1 digit.

So now when your user dials 82 two things will happen. First, the system will recognize 82 as needing to be put on the VOIP trunks. But since you set it to absorb only 1 digit it will pass the 2 on to the VOIP trunks. Since you set 2 to be the code for the VOIP.ms account then the system will know you want to route the call out the VOIP.ms trunk. The final thing is since we told the system to absorb that 2 then your user will now be out on the VOIP.ms trunks without the system passing that 2 to voip.ms. If you hadn't set that 2 to be absorbed then it would become the first digit of the number. Soo instead of sending 1800

http://WWWDELL

to voip.ms you would be sending 21800

http://WWWDELL

which will cause the call to fail.

If you need more help with routing, dial plans, absorbing, etc it may just be better to get a vendor to help. It's a bit confusing.

 

[gberger]

Thanks for sharing.