Connecting to Toshiba PBX - No out dialing

I am trying to install FreePBX on our Toshiba CIX-670 to replace Messaging SiP Voice mail from Toshiba.

I have it working when I send calls to it. All is great. How ever I can not get it to make an out bound call or transfer a caller.

Has anyone gotten it to interface with their Toshiba?

This is my outbound

And my Inbound

This isn’t specificall about your Toshiba, but there are some problems with your Inbound and Outbound Settings:

I’d combine these into a single “friend” trunk (both inbound and outbound, same as “peer&user” if that was a thing) so that the mistakes are easier to fix:


This gets rid of some of the errors in your original settings.

Since both machines are in the local network and (I assume) you are using a firewall, you could change your system to use IP authentication exclusively if the Toshiba supports it. That would get rid of the Username/Password interaction.

You haven’t really specified how the Toshiba and Asterisk are connected to each other, or how you are connecting to make calls, so that’s not really something we are going to be able to help you with.

If your tie-line trunk to the toshiba is local, I would not bother with username/password and make sure your context is “from-internal”. An oldie but goldie is

voicemail and MWI will need your decision as to which system hosts the voicemail, how to so route those callsand how to pass such state changes between the two.

Well, I have tried the setting with no change.
I may not be able to do what I want. I am connecting to SiP stations off of the Toshiba and coming to my FreePBX box (vHost) .

What is working;
Trunk(station on Toshiba) registers
I can call the “station” on the Toshiba side and get my IVR.
If I send a trunk incoming to the Toshiba to the station with a VM Application digits of an IVR Entry it follows and goes to the mail box or IVR fine (normally)
Users can dial in and get their massages.
Users can access the web GUI and get their messages
IP Station(s) Aastra 6757i

What is not working;
Stations can not make calls to Toshiba stations (dial pattern 58XX)
Stations can not make calls out using Toshiba trunks (dial pattern 9|NXXXXXX)

So , from asterisk :-

is your context to the Toshiba trunk “from-local”?
Do you have a route for 58XX to use that trunk?
do you have a route for “9|NXXXXXX” to the same trunk?

From the Toshiba,

Will it route a call to 58XX to it’s local extensiona?
Will it simillarly route calls that match NXXXXXX on that trunk , out through it’s PSTN/SIP trunking?

The first bit you can confirm with the asterisk logs, for the second you will need to use whatever info/call tracing your Toshiba provides, without those logs, then no way to know what is going wrong. Adding SIP debugging might help if and when all the routes are correct

I actually have a Toshiba CIX-670 with a SIP trunk to a FPBX running 14. It is a limited SIP trunk. Based off what you’re describing, you’re setting it up similar to how I setup our SIP trunk between the two.
Toshiba -> Freepbx = All call pass without issue (ie, you can call any extension, IVR, feature, etc)
Freepbx -> Toshiba = Limited by your URI dialing digits (ie, 135 hardcoded extension - basically Toshiba will block any SIP digits not registered in your URI)

This is something I wanted to explore on how to do a true trunk with Toshiba. From everything I’ve read, the Toshiba SIP trunk is treated like a carrier trunk. What I wanted to try and do was setup a “StrataNet” trunk, which would give you true flow of the calls; however, that is proprietary to Toshiba. If you want to post your SIP trunk settings, I can compare it to mine and maybe we can get it working.

When I talked with Sangoma support, they also mentioned setting up a bunch of SIP stations on the Toshiba and have the Freepbx register these extensions on the Freepbx. I haven’t had time to test this, but this is similar to how the old Toshiba voicemail worked. Not sure how you setup the individual stations as a “trunk” in FPBX, but definitely would make sense and allow you to dial any extension on Toshiba. The next hurdle would be how to route ext digit dialing from Toshiba to the FPBX.

Feel free to PM and maybe we can discuss this over the phone as well.


I am new to the forum. I have questions about setting up CIX670 SIP trunking to register on FPBX 14. How do I PM you?
Thank you

I am not a telephony, toshiba or FPBX expert. I can’t answer your questions but I can post what I did.
I browsed for ‘CIX670 SIP trunk’ and found several references for setting up a CIX SIP trunk. Some mentioned when creating an OLG that ‘There is a need to create an OLG access code for this group.’ but no help on how to setup the access code.
I found reference in ‘Strata CIX programming Manual Volume 1’.
Use at your own risk.
I used program 102 Flexible Access Codes to create the access code.
FB00 Access Code: #8 (I am not a telephony expert. I don’t know if #8 is a good choice)
FB01 Feature Name: Line Group access code - one access code for each OLG
FB02 OLG Number: 8 (I had previously created an OLG group 8, service type SIP)

I have this working.
CIX670 SIP trunk will register on FPBX14.
Toshiba -> FPBX. A toshiba station (extension) can call a FPBX extension by pressing intercom and an entering an OLG access code/extension number.
Example: to call (intercom???) FPBX X2270 from Toshiba X250.
Press Intercom button on X250
I get a message in the phone’s display that says ‘DIAL STATION NUMBER OR ACCESS CODE’.
Press #82270
FBPX X2270 starts rings.

What does not work
FPBX extension -> toshiba station. I cannot intercom or call X250 from X2270
FBPX extension -> toshiba CO line. I cannot dial out.

What do you mean by ‘Limited by your URI dialing digits’?
Are you talking about using program ‘309 Direct Inward Dialing’?
What do you mean by ‘SIP digits not registered in your URI’?

What program is used to ‘setting up a bunch of SIP stations on toshiba’?
Has anyone been successful in getting ‘FPBX ext -> CIX ext’ or FPBX ext -> CIX CO line’ to work?

Thank you,

I’ll see about putting a more detailed setup. The URI is located under the SIP Trunking on the Toshiba. By default, it will block all incoming numbers, unless you add them to the URI table. The other issue is caller ID. The best option would be using QSIG, but it sends some odd info in the SIP packet, so Asterisk drops the packet. I’ll post an example tomorrow.

Sorry for the delay, I’m still working on a few tweaking using a PRI trunk to the Toshiba system. I will state the issue I had with the SIP trunk between Toshiba CIX and the FreePBX. As I get time, I would like to make a detailed guide, but here is the brief overview of how I have it setup and the one major issue that had me pursue a PRI trunk using an Adtran instead.

For the SIP trunk to work, you have to first make sure you have SIP trunking license on the Toshiba. Then you’ll also need an MIPU card, which I assume if you’re reading this you already have. In my setup, I then made sure the MIPU had an IP address in the same network as the FreePBX.

On the Toshiba, you need to first create the Incoming and Outgoing Line groups to use in the SIP Trunking. This is under Trunk -> ILG and the Trunk -> OLG. For simplicity, I made mine the same, in this case 41 for both ILG and the OLG.

ILG settings:
Create = 41
01 Group Type: SIP
02 Line Type: Tie
03 Service Type: DIT
04 Private Service Type : Q-SIG
11 DID Digits: 3 (I have 3 digit dialing on the Toshiba system)

OLG settings:
Create = 41
01 Group Type: SIP
02 Trunk Type: Tie/E&M
03 Service Type: Standard (this is where I had issues–see comment at the end)

Next, you have to setup a SIP trunk. This is under IP-Telephony -> SIP Trunking
On the Channel Group Setting tab (Prg 326), click Create and enter a number, again, I chose 41 so everything would match.
01 Equipment: (location of your MIPU card, in this case it was 0301 – ie Cabinet 03 and Slot 01)
02 LAN interface number: 1
03 SIP Trunk Channels: 4 (or however many channels you licensed)
Click Submit

Next, setup your Service Definition (Prg 327)
Again, create and I chose 41
This section is where I didn’t spend a lot of time, so maybe you can tweak things here to make it behave better, but this is what I set:
01 Registration Mode: Client
02 ILG: 41 (or what you used for your ILG)
03 OLG: 41 (or what you used for you OLG)
04 Effective Channel Number: 4 (ie, what you licensed for your SIP trunk channels)
05 Domain Name: IP address of the FreePBX
06 SIP Server: IP address of the FreePBX (believe you can also put :)
35 SIP Trunk Message Option: SIP Server IP Address
36 SIP Trunk Message to Header OPtion: SIP Server IP Address
37 SIP Trunk Register Message From Header Option: IPU IP Address (don’t believe this is critical)
Click submit

Next you have to link the Channel group and Service Definition together, this is under the Service Assignment (Prg 328)
00 Channel Group: Select 41 (ie, what you put in Prg326)
Click on the Service No 1 slot.
Then under 02 Service Index, select 41 (or what you put under Prg327)
Click Set

Lastly, and this is very important, you need to setup your URI. Click on URI
00 SIP URI Trunk Service Index: select 41 (or what you put for Prg326)
Click on the Index 1, which should populate 01 SIP URI Index
Now enter the following data (again, there may be a better way, but this is how I got it working):
02 SIP URI: toshiba
03 SIP URI Username: toshiba
(for this to work, both the URI and username have to be the same)
04 SIP URI password: toshiba (or whatever password you want to use)
05 SIP URI Channel Group: 41 (or what you use in )) SIP URI Trunk Service Index, this should auto populate)
06 SIP URI Attribution: main
Click Add, which is usually next to 00 SIP URI Trunk Service Index

This will then save and populate your 1st index. Now, this is just to enable the trunk, next you have to add a URI for each extension or Multiple Call Group you want users from the FreePBX to call. For example:
I have Ext 111, so in order for users from the FreePBX to call Toshiba, I need to add it to the URI.

Select Index 2, it should populate 01 SIP URI Index with 2
02 SIP URI: 111 (or what extension you want to add)
Leave Username and password blank
05 SIP URI Channel Group: 41 (or what is in 00 SIP URI Trunk Service Index)
06 SIP URI Attribution: sub

Now, you’ve got that setup, now you need to route calls from Toshiba to the FreePBX. For testing, I just added a Flexible Access Code (or System -> Flexible Access Code)
Prg 102 Flexible Access Code
00 Access Code: 87 (or whatever two digit or prefix you want to use to dial to the FreePBX)
01 Feature Name: Line group access code- one access code for each OLG
02 OLG Number: 41 (or whatever you setup for your OLG)

Now, you’re ready to create the trunk on the FreePBX
First, create the Trunk, I did mine with PJSIP, but you can use SIP as well.
Trunk Name: toshiba (this will need to match the URI and username you set for the URI)
pjsip Settings:
Secret: toshiba (or whatever password you used in URI)
Authentication: Inbound
Registration: Receive
Context: from-internal
Codes: ulaw (that’s all I have setup, in Toshiba that is the G.711u found under SIP Trunking Prg 327 - 17 Primary Audio Codec)
Click Submit and Apply Config

Next, you have to setup an Outbound Route on FreePBX. For simplicity, I just created a Prefix, in my case 88, to dial to Toshiba. So create an Outbound route, I called it Toshiba.
Trunk Sequence, choose the Toshiba trunk
Dial Patters: in the Prefix put 88, and “.” (without the quote) for the match pattern.
Click Submit and Apply Config.

At this point, you may need to restart your MIPU card by pushing the reset button. Then try calling. Once I did this, I was able to call between the systems. The downside with this setup is any call coming from Toshiba -> FreePBX will always show the URI or “Toshiba”. So any extension from Toshiba can call Freepbx. Again, for your FreePBX to call Toshiba, you have to add a URI for each extension or multiple call group or Hunt group.

Now, I bet the CallerID would pass across, if you could choose QSIG for the OLG. If you do, as I started with, you’ll see the following in your Asterisk Logs:
[2018-05-27 16:58:22] ERROR[2710]: pjproject:0 <?>: sip_transport.c Error processing 714 bytes packet from UDP :5060 : PJSIP syntax error exception when parsing ‘From’ header on line 4 col 12: PJSIP syntax error exception when parsing ‘Contact’ header on line 8 col 14:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK1ca4f80c9a7462b90f7777b3506346c8
Max-Forwards: 70
From: <sip:▒[email protected];tag=6955b0ae3ecb2293
To: sip:[email protected]
Call-ID: [email protected]<MIPU IP Address
Contact: sip:▒[email protected]<MIPU IP Address
Supported: 100rel
Content-Type: application/sdp
Content-Length: 293

Notice the ▒, not sure what this is, but basically Asterisk sees it and throws an error. If there was a way to filter that out, I’m sure it would work. My guess is it is a Toshiba StrataNet specific packet that is being sent. Hence, it won’t work. Because of this issue, I’m now using a PRI card (BPTU) card with an Adtran 924. This solved the CallerID issue and allows calling between the system. I’m still working out some other issues, but so far, much better solution.

So in short:
Toshiba to FreePBX SIP trunk is possible, but limited, just need to figure out how to get QSIG working
Purchase a BPTU (PRI card) for the Toshiba and use a device to translate to SIP, such as an Adtran 924

As I get more time, I’ll see about putting together a better guide and how we setup the PRI trunk.

If anyone has any ideas on the QSIG or better options, feel free to comment.

The more I learn about connecting a CIX670 to FPBX, the less I think I know.
I followed your instructions except with these differences

ILG settings (prg 304):
Create = 8
01 Group Type: SIP
02 Line Type: CO (I did not use TIE because I don’t know what a tie line is)
03 Service Type: DID (I did not use DIT because I don’t know difference between DID and DIT)
04 Private Service Type : Standard
11 DID Digits: 3 (I also have 3 digit dialing on the Toshiba system)

Can you explain difference between CO and TIE?

OLG settings (prg 306):
Create = 8
01 Group Type: SIP
02 Trunk Type: CO/DID (I did not use TIE/E&M because I don’t know what tie or E&M are)
03 Service Type: Standard

Can you explain difference between CO/DID and TIE/E&M?

SIP trunk (prg 326):
Create = 8
01 Equipment: (location of your MIPU card, in this case it was 0301 – ie Cabinet 03 and Slot 01)
02 LAN interface number: 1
03 SIP Trunk Channels: 1 (number of purchansed SIP trunk channel license - see note-1 below)
Click Submit

Service Definition (prg 327):
Create = 1
01 Registration Mode: Client
02 ILG: 8 (or what you used for your ILG)
03 OLG: 8 (or what you used for you OLG)
04 Effective Channel Number: 4 (ie, what you licensed for your SIP trunk channels - see note-1 below)
05 Domain Name: IP address of the FreePBX
06 SIP Server: IP address of the FreePBX (believe you can also put :slight_smile:
35 SIP Trunk Message Option: SIP Server IP Address
36 SIP Trunk Message to Header OPtion: not available on my version of CIX firmware
37 SIP Trunk Register Message From Header Option: not available on my version of CIX firmware
Click submit

Note-1: I only purchased one SIP Trunk Channel license. I found that I could only have one simultanious conversation accoss the SIP trunk.
prg 326 - FB02: 1
prg 327 - FB04: 4 (this limits the number of simultanious conversation, but you must have a SIP trunk license for each conversation)
I could intercom from FPBX X1157 to CIX X144 and start a conversation with no problem. While X1157 -> X144, I would try to intercom FPBX X1144 -> CIX X157. On X1144 I would hear ‘The number you dialed is unavailable’.

SIP URI Assignment (prg 329):
Select Index 2, it should populate 01 SIP URI Index with 2
02 SIP URI: 157 (or what extension you want to add)
03 SIP URI User Name: 157
04 SIP URI Password: password
05 SIP URI Channel Group: 8 (or what is in 00 SIP URI Trunk Service Index)
06 SIP URI Attribution: not available on my version of CIX firmware

I entered user name and password because I created an FPBX Extension for the URI to log into. I found that the CIX 157 URI would register on FPBX. I also setup a FPBX Outbound route. After 157 registered then I was able to intercom from FPBX to CIX.

What is working
toshiba station -> FBPX extension. I cannot intercom or call X2270 from X250
FPBX extension -> toshiba station. I cannot intercom or call X250 from X2270

What does not work
FBPX extension -> toshiba CO line. I cannot dial out.

Does anyone know what access code/dial pattern the the FPBX needs to send to the toshiba to allow the call to be routed from the SIP trunk to the analog CO line?

Thank you,

Off the top of my head you may need to enable trunk to trunk transfers in that Toshiba’s cos for that sip trunk (I would make it a cos that you don’t use for anything else as that is not a good idea to turn that on for normal trunks). You may have to setup DISA on the Toshiba in order to make that work.

The reason I went with the TIE setting was to allow all outbound calling from the Toshiba system. This is based off the setup of the IPEdge (Toshiba’s IP based phone system). It uses StrataNet and setup the trunks as TIE. In order to change the inbound and outbound group, you’ll have to create new Inbound / Outbound routes or delete your SIP trunk and then change them.

Based off my experience, if you really want to have a true trunk, allowing ext to ext calling with Caller ID between the two system, go with a 24-channel PRI card and then use a PRI gateway to connect it to your FreePBX. So far, that has been working well with our system.

The more I work with the SIP trunk, the more I understand it limitations. As I learn more, I go back and re-read this subject. Your comments about a BTPU PRI card and Adtran24 make sense.
Is ISDN PRI the same as T1?
I notice the BTPU has two ports on the card edge. I am guessing one of them is the 24 channel T1 port. I am guessing the Adtran 924 is a T1 to SIP channel converter.
Is that correct?
Is the connection between the BTPU and A924 a single cat-5 cable with RJ-45 ends?

Thank you,

The PRI is a T1. It will be the 2nd digital interface on the Adtran. And yes, it uses a Cat5 straight-through patch cable. I’ll see about posting a config file example later.

Technically they use RJ48 and NOT RJ45, the physical connector looks the same but the pairs are different and cat-5 is not in spec for either T1 nor PRI(US) but you won’t notice if the length is less than a few hundred feet

Google will show you the different pairs used on a “turn-around” on a DS1 or Ethernet

What toshiba CIX license(s) are needed for the BTPU card and 24 channels? I don’t think I would need all 24 channels. I can see using 12 channels to cover possibility of all 6 toshiba CO lines and 6 intercomm being in use at same time. Do you know if all 24 channels are covered under one license or is each channel covered by an individual license?

I will check out the RJ48/T1 wiring. I have RJ11 and RJ45 crimping tools and made cat-5 straight and twisted (crossover) cables.
Thank you,

It just seems to me that if I can get it to answer SiP “Stations” off of the Toshiba fine there should be a way to mess with the dial plan to get it to call out

@jakthree PRI’s use basic licenses on the CIX. You will need one basic license for each B channel you configure. If you use 12 channels on the PRI you will only need 12 basic licenses as you will only setup 12 B channels. If you have more basic licenses available on the CIX I would just go ahead and setup a full 23 channel PRI. A full PRI is comprised of 23 B channels (voice channels) and 1 D channel (a call control for all the B channels). I can’t remember if the D channel requires a basic license in the CIX as well. My guess is that it probably does but it has been a couple of years since I’ve managed a CIX.

Interesting, I talked with the company that supports our Toshiba system and they said to just use a straight through patch cable. Looking at the pinout, it looks like it should be the same. Maybe I’m looking at the wrong pinout?

Or if it is this way, then the pair are just switched around:

Not sure if it would make a huge difference, but in this case, it is about 5 ft and is working fine. The only issues I’ve experienced so far are:

  1. Unable to call to the IPEdge trunk (ie, it won’t route to the Stratanet) Freepbx -> CIX -> IPedge doesn’t work or vice versa.
  2. Blind transfers have one-way audio if you transfer to a Ring Group on the FreePBX. Using Queues or Follow-me doesn’t cause an issue. Also the Toshiba Ring back timer will pull the call back if calling a Ring Group as well. Believe it has something to do with who takes over the call, so to speak.

My end goal is to see about having the Toshiba call out through the FreePBX.

I’m also trying to work through these two issues to fully offload most of the functions of the Toshiba to the FreePBX:

  1. Seeing if I can direct the CIX extensions to the FreePBX (in my case, PBXAct) voicemail. (ie, offload voicemail to FreePBX)
  2. Paging Toshiba phones through FreePBX. This is the part that is tricky, since they are not extension on the FreePBX, I don’t have a way to call them. That might be a different thread for this. My thought right now is to see if there is a way to have the FreePBX register a SIP extension on the Toshiba, and use that for a trunk, so to speak. Ie, you call out that specific trunk for paging.

I’m not sure on the licensing, as my CIX is showing the T1 card using StrataNet licenses. With the IPEdge, we bought the StrataNet Unlimited license. I know the IPEdge is using 12 channels and the T1 card must be using 23, as I show 35 used in StrataNet. Most likely, this is due to the Inbound/Outbound using Q-SIG.

So in terms of licensing, if you go with T1 card, in my case, it is using StrataNet licenses.