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Connecting to Toshiba PBX - No out dialing

(christrati) #21

In terms of registering, I’m referring to the SIP trunking and the URI tab.

  • IP Telephony -> SIP Trunking

  • Click on the URI tab

  • Select your Service Index

Here’s a screenshot from my Toshiba system on that SIP trunk (which I’m no longer using):


For complete access to your dial plan like paging et al. , then your tie-line context will need to be from-internal , incoming calls from the pstn should probably use from-did-direct to facilitate the one-to-one mapping of extensions between systems.

If the link is up then the wiring is fine, just check for any clock slips or framing errors in any hardware diagnostics available, Debugging the sip sessions and the ensuing sdp traffic would be my next step as to missing media.

(christrati) #23

I have the FreePBX trunk to the Adtran setup as a from-internal context. I setup my Outbound Route to the Toshiba specifying the extensions and I put a “88” as a prefix and then “.”, so I can dial “88” and then any number on the Toshiba. The issue is any of the “Feature Codes” on the Toshiba don’t work. We have the same issue with the IPEdge, which is why we never fully implemented the system. We couldn’t page both systems at the same time. I may have the same problem with FreePBX, as I need a way to page on both systems. And if there is a feature code that requires an additional prompt, the FreePBX will state it doesn’t answer.

What I did get working is if I set a Feature Code, but made it like an extension. For example, if I dial 88403 (Feature code 403 on the Toshiba), it will do an All Page only on the Toshiba system. Now, I’m not sure how I can have FreePBX call both 88403 and a page group on the FreePBX.

In terms of the link, I’ll have to see if there is something on the Toshiba system to monitor, or I’ll look at the logs with the Adtran. So far, I haven’t any call issues.

Thanks for everyone responding to this thread, hopefully this will be helpful for anyone trying to tie a FreePBX to a Toshiba system.


maybe a hint or two at

(christrati) #25

I’ve been looking over that thread for quite some time. The difference is that I’m using an Adtran 924e between the FreePBX and the Toshiba CIX. Most likely, I’m just misunderstanding some of the custom config and how to route to to the FreePBX voicemail.

Trying to map the post to match my Adtran config, I see it going to a custom context of vm_from_toshiba. Looks like the setup example was going to a T1 card in the server. My guess is I need to do some tweaking on the Adtran or see about using that custom context. Currently, I just have the Adtran direct it straight to the FreePBX and I use the from_internal context.

I do like how the article has a placeholder for the Toshiba Strata CTX670 (ie, CIX670) and Asterisk. Maybe this thread will help with that.

Since the end goal is to move the Toshiba ext to the FreePBX, I was thinking about doing the following:

  • Create a new extension on FreePBX with Voicemail
  • Add a SIP alias to the FreePBX ext of the original Toshiba Ext
  • On the Toshiba, change the DN of the Ext to a different number
  • Change the Display DN on the Toshiba to show the original Ext number
  • Remove the System Call forward on the Toshiba Ext
  • On the Toshiba, add the original ext to the Strata Net DNs, sending it to the FreePBX
  • Lastly, on the FreePBX, enable Follow-Me and have it ring the renamed Toshiba Ext
    The downside is there is no message notification, but it would essentially function the same, but will use up two channels whenever someone calls that extension, as it will take up 2 channels, as one will be dialing to the FreePBX and the other channel with the FreePBX calling the renamed Toshiba ext.

In terms of other references, I’ve looked at the following as well for the integration:


You might well find RBS T1 signaling using E&M is more than sufficient for your tie line over fighting your way through ISDN and qsig.

(Rich C) #27

I am in the process of following and looking into this thread. I am personally looking at connecting my Switchvox to our CIX-670. I have some questions with regards to what worked and what isn’t.

  1. Above you stated that extension to extension worked but then beside it it states that ext to ext dialing didn’t work…with your config did ext to ext dialing work?

for right now at least I have zero interest in having the SVox (or in your case FPBX) call out via my Toshiba PBX. So I am quite alright if that feature doesn’t work. Caveat being so long at it doesn’t generate so many errors that both systems slow to a grinding halt…

I too am trying to find the different in the Trunk setup between CO/DID and TIE/E&M. If I find the detail difference between the two I will post here. If you have already found out what the difference is and which works best - I’d be grateful if you could post that as well.


(Rich C) #28

I have heard/read that the Toshiba’s will use a (I believe it was called an LSR…?) Either way what that string likely has at the beginning is the Toshiba node. When I look at your log I see Node+Prefix+ext. I could be off and be completely wrong but I recall reading that somewhere.

(christrati) #29

You’re correct, the Toshiba with use a Node ID, which is part of StrataNet. I believe the main question is how you can trunk them together. I have it setup 3 different ways, and each have their pros and cons.

  1. T1 - PRI - This works well for Ext-Ext, but I’m unable to dial out the Toshiba. Ie, FreePBX -> Toshiba -> Trunk gets blocked. For this setup, I used a TIE trunk and QSIG.
  2. SIP Trunk - Works ok for Toshiba to call to extensions, but FreePBX -> Toshiba will limit inbound calls by URI. In other words, you restricted to ~150 numbers that are allowed inbound through the SIP trunk. This one I had to use standard SIP for the outbound; otherwise StrataNet send an extra packet in SIP and it causes Asterisk to drop the packet
  3. Analog - Basically, I use a bank of Analog extensions (RSTU) cards in the Toshiba and a Sangoma Vega 60 (FXO ports) and use that for the trunk. This is how I fixed dialing certain feature codes (still can’t use any with #, but everything else works). My main issue was paging and MWI on Toshiba phones. Paging works great, and the MWI works, but I just haven’t figured out a good script to handle turning on and off the voicemail light.

In terms of the Node ID, that is all part of Toshiba’s StrataNet. And you can use LCR routes to strip off the node when sending it out. On your CIX, you can go to StrataNet -> Route Choice. This is where you can set a Strata Net route to go out a specific OLG and then use the Digit Modification Table to delete the leading digits:

Here’s how mine is setup.

(Rich C) #30

That is perfect for me. See we have three sites which use the Strata 670 and that would mean that a SIP trunk would need to be established at each of those sites. In your example I’d simply use:

Office 1
Office 2
Office 3
As my name and UID instead of your example of using ‘toshiba’. This way the inbound would be more descriptive. What is also a good thing is that each of those offices will never have a headcount over 60 people. So the ~150 URI limit is actually not a limit for me at all. Although thank you for the information. Just to confirm using SIP trunks were you ever able to get proper CID to work? I am hoping to get this all tested early next week. I will post here what worked and what didn’t but because I am not using FPBX I don’t want to “muddy” the forums with settings on the Switchvox side. I will however post any settings I can find which I can confirm work to address that lead packet.

As an aside did you ever try and run a Wireshark capture to see if you could read that character found in the quoted text below?

Contact: sip:▒88111@<MIPU IP Address


The post should read.

What is working
toshiba station -> FBPX extension. I can intercom or call X2270 from X250
FPBX extension -> toshiba station. I can intercom or call X250 from X2270

I am sorry for the confusion. I have dropped the attempt to connect FPBX to CIX-670 when I found that you have to have a Strata SIP trunk license for each concurrent conversation between the FPBX and CIX. SIP trunk license are to expensive. The FPBX to CIX project has been put on hold due to the company decided to purchase more Strata IP phones and licenses.
Thank you,

(christrati) #32

With SIP trunks, and setting it up for TIE on the Incoming trunk, I would get Call ID fine. But outgoing you were restricted to whatever you set for your URI. That was the big reason I went with the PRI setup, as Caller-ID for internal calls would flow correctly. I still have some issues with Caller-ID between the system, mainly with how transfers were and caller ID not updating. Basically, the Toshiba doesn’t really have an immediate “blind” transfer. All the phone’s have a CNF/TRF button, and when you hit it, it basically does an attended transfer. Now, once you hang up on a Toshiba-Toshiba call, it will update, but to the other trunks, it doesn’t update.

And in terms of a Wireshark capure, no I haven’t really done any additional testing, as I didn’t know of a way to remove that character and it was pretty much a show stopper for using the TIE option. More than likely, it is some proprietary packet Toshiba is adding in for their StrataNet.

For licensing, if you go with SIP trunks, it does require you to purchase SIP trunk channels. If you go with PRI, it hit against my StrataNet licensing, which I bought an unlimited licenses for my CIX.

(Rich C) #33

Great information thank you. If I recall from earlier in the thread you’re connecting PBX’s which are at the same location (physically). My case I have a total of 4 PBX’s at 4 different locations (52 KM/32 miles) apart but all connected on the same LAN network. Today is my testing day so we will see how it goes.

(Rich C) #34

So I have setup the following:

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)

NOTE There are a ton of other default settings which were left alone.

OLG settings:
Create = 41
01 Group Type: SIP
02 Trunk Type: Tie/E&M
03 Service Type: Standard

NOTE There are a bunch of other settings there which I left as default.

SIP Trunk:
ID: 3
01 Equipment: 0206
02 LAN interface number: 1
03 SIP Trunk Channels: 4 (Licensed for 6)

Service Def:
ID: 1
01 Registration Mode: Client
02 ILG: 41 (same as above)
03 OLG: 41 (same as above)
04 Effective Channel Number: 4 (licensed for 6 testing 4…not sure if that is my problem)
05 Domain Name: /
06 SIP Server: /
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
38 SIP Trunk Register Message To Header Option: SIP Server IP Address (other option is FQDN) - I am not sure if this is configured properly

Service Assignment:
00 Channel Group: 3
02 Service Index: 1


00 SIP URI Trunk Service Index: 1
01 SIP URI Index: 1
02 SIP URI: MandPOffice
03 SIP URI Username: MandPOffice
04 SIP URI password:
05 SIP URI Channel Group: 3
06 SIP URI Attribution: main
Then clicked add.

Other URI Entry (just for testing):
01 SIP URI Index: 2
02 SIP URI: 266
03 SIP URI Username: blank
04 SIP URI password: blank
05 SIP URI Channel Group: 3
06 SIP URI Attribution: sub

NOTE The 266 above is an extension from the Toshiba system and does NOT exist in the Switchvox/FreePBX.

Toshiba Flex access code:
00 Access Code: 87
01 Feature Name: Line group access code- one access code for each OLG
02 OLG Number: 41 (as entered above)

Now this is where I differ from…well everyone here. Enter Switchvox land…:
(I see no where to enter PJSIP) - so I enter in a VOIP provider:

UID: same as SIP above
PWD: same as SIP above
IP: of the MIPU card lets just call it (
Then I set it to peer rather than provider in the settings.

Now in the server side it is showing SIP connected and “working”. I then also added and inbound and outbound call rule on my Switchvox.

If ext 266 is dialed then send call over .

If ANY extensions on Switchvox come in on then route to that extension AFTER trimming the first tow digits

NOTE I was not sure if the leading twe digits get sent or not (ie. the felx access code digits of 87).

My results:

  1. SIP is connecting between Switchvox & Toshiba
  2. Switchvox is NOT able to dial to the Toshiba (error 85 - refers to any failed call over SIP)
  • ISDN Error: Received status of CHANUNAVAIL with a cause code of 21
  1. Toshiba calling Switchvox recieves a busy error

I noted:

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.

All indicators are that this is likely an issue with the Toshiba and NOT the Switchvox. Any thoughts?

(christrati) #35

I will say I did demo Switchvox and I had issues getting the SIP trunk to work at all, which is why I went with FreePBX. I wasn’t sure how to add it in order for it to work correctly.

In terms of dialing, are you dialing from Toshiba -> Switchvox by dialing 87+Ext?

In terms of Switchvox dialing to Toshiba, is there any options to set the context? from-internal?

(Rich C) #37

The SIP is showing connected on Switchvox but it appears based on the logs that the Toshiba is blocking the call. The Toshiba is acking the SIP call from Switchvox and then rejecting it. I am wondering is this might be an issue of the extensions internally not “seeing” a proper path to SIP trunk… (on the Toshiba side)

Toshiba’s calls when dialing 87+ext are not working at all. In fact their call is not even hitting the Switchvox.

I have also noted the following error in the switchvox logs:

res_pjsip_registrar.c: AOR ‘MandPOffice’ not found for endpoint ‘sip_provider_104’

Oh and for the record I too prefer FreePBX :slight_smile:

(Rich C) #38

I have noted in the screencap that the UID & PWD are present for the 1st ext with the type “sub”. One I assume that the entry in Index 1 and entry in index 4 both shared the same UID and PWD and two was this critical to get the SIP connection working?

2nd who do you use for Toshiba support…I wouldn’t mind reach out to them to see if my organization cannot finally get this done once and for all.

(christrati) #39

On my setup, I only had to put it for the main, I didn’t have to put it for the subs. They did all use the same username and password. For your Switchvox trunk, do you have it setup with a username and password for the SIP trunk?

Like I said, the setup I described here should work fine for FreePBX. Switchvox, I couldn’t get working at all with our CIX-670.

(Rich C) #41

Yes the SIP trunk requires a UID and PWD.

Also of note - I very much wish the Switchvox forums had such an active and interactive group…I expect that I will be getting to the bottom of this in the next day or so. I will post my findings. If but for nothing else to prevent others from going through this.

(Rich C) #42


I just wanted to thank you for all of your time on this with me. We have gotten inter system extension dialing working.

WOW what a pain in the rear.

For the benefit of the rest of the world I will be posting details bot here and in the Digium forums of how I got it working and what settings I ticked and un-ticked to get it working. Ultimately it is just a minor variation on the settings you posted in your detailed post. If I were to summarize - Toshiba needs an AUTH whereas Switchvox does not.