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BCM50/FreePBX integration Sip or H.323

AShammah

Vendor
Sep 7, 2023
51
US
Hey all,

I'm trying to connect a BCM50 to a FreePBX instance via sip or H.323. So I can get SIP phone functionality on the BCM50 for Sip Video doorbells.
I'm looking to get the doorphones on the freepbx to ring the current doorphone hunt group 474 on the BCM.

When I try to connect sip BCM says "forbidden from 192.168.1.245:5060" which is the FreePBX.

H.323 is setup but doesnt seem to work.

Hitting the doorphone button set to call 474 I get a Freepbx message "all circuits are busy now, try your call again later"
Dial pattern is set to X. on sip and H323/192.168.3.2/${EXTEN} on H.323 on the Freepbx.
 
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This is how I have my target lines setup:
Voip.ms trunk calls coming in on Target lines 125-127 get routed to separate IVR's then to the appropriate extension or hunt group.
Calls coming in from freePBX target lines 128-129 are added to the current BST doorphone hunt group 474 set to ring broadcast on all extensions.

So I would need to create an extension for each door phone or any other type of phone thats on the freepbx in the BCM and set the target line as that DN # under line type to get BCM dial tone. (y)


Yeah I was thinking that would work Hik - Panasonic :)


1731812437939.png
 
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@daniel05

I cant get freepbx to register with the nortel using your settings.

I get not found from 192.168.1.245:5060 which is the FreePBX

1731819465406.png
 
1731858915547.png1731858964140.png
I have my dial patterns set up as 2xxx for BCM digital, 3xxx BCM IP Phones, 4xxx BCM hunt groups
1731858996499.png

I tried using Outbound, Inbound, Send & receive. I still get not found from 192.168.1.245:5060 on the BCM
 
Got the trunk registered :). Doorphones work perfect (y)
I had it calling BCM-FreePBX when it was on "Public" using 89 to dial to FreePBX. I removed the public now I have "89" as dest digits in the private routing table it doesnt call out to the FreePBX.
I also still don't understand how to set DN's in target lines to give FreePBX extensions BCM dial tone so they can access BCM Line pools *codes ect.
I have a few 8865 Cisco's I'm going to convert to MPP. I'd like to see if they display the Hikvision Doorphone video feed on a call.

1732167989005.png

1732170838574.png
 
Hello everyone

I've finally got my BCM 50 to work with my Raspberry pi 3 on FreePBX last week.

I have it working fine using a linked numbering scheme.

The only thing I'm having problems with is getting the caller ID to work from the FreePBX to the BCM. It only displays the Trunk caller ID.

My understanding is that this only applies when the extension caller ID isn't available.

Has anyone else had this issue?
 
Hey Firebird,

Thats great!!! All you have to do is delete the call ID from the outbound route and trunk in FreePBX.
Let the extensions themselves forward the CID info eg. Front Desk <401> Sends the name First.

I ordered a Pi4. which version FreePBX and what did you use to install it?
I also ordered a cheap Lenovo i5 to put ESXi on incase that doesn't work out for me.

My trunks are connected. I have ring group 480 dialing target line 129 then added target line 129 to Hunt group 474 for the BST doorphones. That all works great.
I'm still having trouble linking the two systems together to get each others dial tone. I was able to dial from BCM to Freepbx at one point but thats as far as I got with that.


This was the closest guide I could find. But I'm still not understanding how it works.

I think I may end up going @daniel0581 route and have FreePBX handle everything (PSTN trunks) and let the BCM slave. Maybe just keeping the IVR at the BCM. For now.

Converting my Cisco 8865's to MPP, I'm waiting on the migration licenses. I want to see if I can get the Hikvision doorphone video feed to come up on the display.
 
"I'm still having trouble linking the two systems together to get each others dial tone."

This is how I see it...

The sets do not "register" to the other PBX therefore no features etc, only the SIP Trunk accounts register, to create a com's link.
You have a setup to call the other sites' DN's using Target Lines & Routes over a SIP connection as if an internal call, that's all!

Think of your setup as this, that I did for fun:

Women Telephone -SIP3.jpg


You would need proprietary Nortel IP sets to have the full access to features or a connection to an ATA port for * codes as mentioned.
On the BCM, Nortel was just about finished the SIP set programming they started on the backend, but they went bankrupt.

Maybe DISA would help in small scenario.

A Target Line points the DN of an analog set/port which is connected/wired to a trunk port which is set to "auto answer with DISA".
 
Hahaha I like that diagram :)

All they had to do was finish that one last bit and the BCM would live forever hahaha.

I was thinking DISA. I was trying to get around DISA not working with VOIP.
I'll try that and see if it works. Thanks (y)

Its a small home system 20 ext's only 5 T7316E's the rest are IP nortels, the 2 BST's coming down this week. Being nostalgic or else the BCM would be in the closet with the CICS. I'm not ready to retire Nortel Nancy yet lol
 
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I've been trying to get DISA auto DN without Cos to work for a bit.

Not exactly sure where to route what. It's a little confusing.

Bellow as per the BCM manual this is a tandem call but I'm not seeing detailed instructions on how to do it.

"Answer with DISA cannot be administer to voice over IP (VoIP), since they do not connect
systems outside the private network. However, a user calling in remotely on another system on
the network can use the trunk to access the system or a user calling in on a PSTN line can use
the trunk to access the private network. To provide control for this type of access, ensure that
you specify remote access packages for the trunk. This type of call is called a tandem call."

"VoIP note: You can also use VoIP trunks between some or all of the nodes. The setup is the same,
except that you need to create gateway records for each end of the trunk, and routing tables to
accommodate the gateway codes, or you can configure a gatekeeper. Refer to
interoperability: Gatekeeper configuration" on page
Chapter 2 System telephony networking overview
389"


Page 389 looks like CS1000 interoperability....
 
Update: If I dial 9+mobile number from the Freepbx Cisco 8865 will dial out on the Nortel trunks.
Also will dial from any Nortel trunk if i put in the trunk code eg. 82,85,87. Those are seperate phone numbers for private lines.

Still no DISA
 
"A Target Line points the DN of an analog set/port which is connected/wired to a trunk port which is set to "auto answer with DISA"."

It took me awhile to sort out too so I had made a cheat sheet:

To dial in remotely to page sets or dial out on a trunk.
Those are the only 2 choices.

BCM wiring using spare ATA then back into spare trunk port (loop spare Trunk port 064 to spare Analog Station port)

In this example we are using:
Spare Analog port DN 444
Trunk port Line 064
Call Security Restriction Set filter 05 (& or Line Filter 05) , Remote Access Package 05, COS 05

3 answering options depending on your setup:

Option 1 - Auto Attendant answers
-Set AA to answer your spare Trunk Line port on 0 rings
-If you use this option you then dial the DN of the spare analog port 444 after AA answers, you will then get DISA dial tone to enter password


Option 2 - Trunk port answers
-Assign the trunk port 064 to ring at the spare analog port 444
-You will then get DISA dial tone to enter password

Option 3 - None of the above, Intercom the analog DN 444
-A networked remote user would intercom the spare analog port 444, you will then get DISA dial tone to enter password
---

Lines/Active/064/Properties/
-Trunk Mode = Supervised

Lines/Active/064/Preferences/
- Auto Answer, DISA =Yes

Filter 05 for Set &/or Line
-If you allow Pool or BlocX access then restrict any digits required such as 0 for overseas

CallSecurity/Remote Packages
-Package 05 - Remote Page = Yes, and add Pool and/or BlocX if allowing to dial back out

CallSecurity/ClassOfService/05
-Enter 6 digit password, restriction filter 05, remote package 05

Sets/444/Line Access/Line Pool Access/
-add your line pool/s access if you want to call back out

Sets/444/Capabilities/Receive
-Short Tones = Yes, this will allow dial tone to be broken when DISA answers.

Sets/444/Capabilities/ATA Settings/
Disconnect Supervision = Yes


Speed Dial/Auto Dial - dialing string:

For AA option:
XXX-XXX-XXXXP444PPXXXXXXP*6XY


Explanation of string in order of appearance (you may need to play with the pauses):
First set of X's = Phone/DID number
P = One pause (or one comma on mobile phones) based on AA answering on 1 ring
444 = DN of the spare analog port
PP = Two pause's (or two commas) to wait for dial tone (You can try the D option instead when waiting for dial tone)
Second set of X's = COS password
P = one pause (or one comma)
* = Feature access (paging only)
6 = Page code
X = Sets or Speakers or Both ( 1 for Sets, 2 for Speakers or 3 for Both Sets & Speakers)
Y = Zone # (0 for all zones, 1 for Zone 1, 2 for Zone 2 etc.)

For direct trunk access via phone number/DID
XXX-XXX-XXXXPPXXXXXXP*6XY

For networked user option
X444PPXXXXXXP*6XY
Where the first X is the other PBX's access/dest code

---

I have just tested all 3 options successfully using only a VoIP Trunk DID with Voip.MS.
I have not tested PBX to PBX but I cant see why it would not work.

Let me know if something does not work or needs editing.
 
Hey thanks for that!

I changed 061 to auto answer DISA after that I cant call in to any of the voip.ms trunk lines. I managed to get one to work by reentering the target line pub rec #
then when I would call in I would get something like "the command you entered is not valid" then AA as usual.

I turned off AA DISA on 061 and I'm still getting "the command is not valid" prior to AA on VOIP.ms trunk line.

Still can't call in to my Toronto trunk (wifes line) or my business line trunk. Only the main house # works.

I have 1 line thats coming in from FIOS on 064. That one works like normal, hits its own AA.

I still don't understand how I can dial 9+XXX-XXX-XXXX and go through any of the Nortel trunks to outside from FreePBX hahaha.
If I can dial 9 and have it understood by the Nortel trunk cant I have a DN # or FT code number send the same way?

I'll see if I can get this sorted out later tonight.

PS: No video feed from Hikvision doorphones to Cisco 8865 3PCC. Cisco will turn on video but nothing is sent from the doorphone.
 
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Now having an issue where if I call in to the voip.ms trunks it hangs then disconnects no ring.
If i use the trunk to make an outgoing call I'm able to make inbound calls to the trunk for a few minutes then its back to no incoming calls across all trunks.

I have to make an outgoing call from each trunk to get each one to accept incoming for the few minutes.
 
This post just saved me from resetting the BCM and starting over from scratch.


Changed ITSP “from header proxy address match” didn’t help.

I turned signaling method on to “options/30”
to give it a tap It’s allowing inbounds now without me having to wake it up.

Not sure what caused it. Everything related to DISA was turned off and I reset the BCM.

Weekly backups are now turned on.

I’m thinking I may just go “green box” for the simplicity of a Nortel/Asterisk coexistence.
 
Well...I am highly confused!
It's best to keep different issues in different threads.

Is everything working then?

Is DISA working?
Are the Door Phones working?
Are all Voip Trunks working?
Are all all internal calls working between the systems?

"I’m thinking I may just go “green box”
What does that mean?

Also note the FAQ's button up top where there is some usefull info.
Here is some SIP info at the bottom of page 3.

Cheers
cc
 
Sorry for the confusion.

Is DISA working?: DISA is not working. kept getting DISA invalid entry then AA played after. DISA is off now.
Are the Door Phones working?: Doorphones are working
Are all Voip Trunks working?: All trunks are working now that I set Nat pinhole options to 30, slight delay now when ringing in to the system over voip.ms.
Are all all internal calls working between the systems?: Internal im only able to call to FreePBX extensions using a dial code and route. I haven't been able to call from Nortel-FreePBX yet.

"I’m thinking I may just go “green box”
What does that mean?: The new E-metrotel Galaxy Pro - Setup with DSM16 and FXO/FXS card's.
 
That's interesting!. I've just spent a couple of days on and off tinkering with the Raspberry pi as I had lost my SIP link and the restore I have scheduled every week didn't work.

I've now got my BCM to FreePBX online and can easily call FreePBX to BCM OK, but I'm struggling to get any BCM to successfully ring a FreePBX phone.

I managed to get one working, but other FreePBX DID's won't ring.

Is it possible to have a few screen shots from the FreePBX please?. I'm certain my BCM is OK.

What I have are 39xx on the BCM and 49XX on the FreePBX.

I just wanted to dial each site without the need for any access code.
 
Sorry for not responding as I've been preoccupied the past couple of weeks and haven't been on here much.

@Firebird Scrambler do you have the context on the sip trunk in FreePBX set to "from-internal"? If not, change to that and try making a call from your BCM to your FreePBX.
 

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