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
URI:
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 (192.168.0.43)
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.
Outbound:
(paraphrase)
If ext 266 is dialed then send call over .
Inbound:
(paraphrase)
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:
- SIP is connecting between Switchvox & Toshiba
- 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
- 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?