It's easy to do this with the FXS ports on the TA908e.
1. Create a new voice trunk type SIP pointing to the FAX SIP server. For example name it T11.
2. Create a voice user with the DID of the fax machine. Under that voice user do the following:
- Create a sip-identity of the username of the fax on the SIP server and specify the trunk, for example T11.
- Enter the SIP password under sip-authentication.
- Enable modem-passthrough
- Disable call-waiting, PLC, and echo cancellation
- Enable T38 with appropriate settings
- Connect it to the FXS interface, typically fxs 0/1 for the first instance.
If you're only connecting one or two analog ports, this gizmo (telco slang "harmonica") simplifies the physical wiring to the Adtran VOICE jack: Allen Tel AT153AM-01 Modular Adapter-25 Pair Ribbon | Graybar Store Available at Graybar locations or online.
I set this up as you advised, but now there are dialing issues with the fxs ports.
Luckily, the in-service PRI is working as it was.
Here is the config (sanitized)
voice trunk T11 type sip
sip-server primary 192.168.128.195
codec-list SIP-Codec both
The 2nd of 3 voice users (fxs):
voice user 5555555555
connect fxs 0/2
forward-disconnect delay 250
sip-identity 158152-fax001 T11 register auth-name "158152-fax001" password "1234VXh84321"
These are the only added lines to support the fxs users. They are registered on the sip server.
1. Any calls that the fxs ports make go to the PRI; none go out to the sip-server primary @ 192.168.128.195.
2. Calls that should be going to 5555555555 / fxs 0/2 are ending up at the PRI as well. They do not get mapped to the fxs port / voice user.
What needs to be added so that 5555555555 routes to fxs 0/2, and for all calls that fxs 0/2 make go to the SIP server?
You'll probably need to configure SABR to steer the calls as suggested. On the SIP server for the fax you'll need to creat a voice grouped-trunk to accept all calls, or at least the patterns that you want the fax to be able to dial.
See the following guide for SABR: https://supportforums.adtran.com/servlet/JiveServlet/downloadBody/1862-102-2-1978/Source%20and%20ANI%20Based%20Routing%2…
The voice-grouped trunk was a key object. I have it working inbound to the fax now. Here's what I did -
1. Added the following to the config:
voice dial-plan 1 extensions 5555555555
voice grouped-trunk FAX
accept $ cost 0
2. In my switch (asterisk), I changed the extension from regular sip user (username 158152-fax001) into a 'trunking' extension, and forced a DID to it, 5555555555. In debug voice verbose, I had noticed that 158152-fax001 was getting delivered previously, and there doesn't appear to be any way in the TA to translate any alpha-numerics or hyphens. So I forced it to the same DID as the voice user 'name' and it has started working.
3. For outbound, hopefully the voice-grouped-trunk will solve that as well, and added this dial plan:
voice dial-plan 2 local NXXXXXXXXX
I will update later when I can get someone remote to test outbound.
If you have different SIP servers for outbound fax vs. outbound voice calls, you'll need to configure SABR for them to route correctly. With two trunks configured to accept "$" or a matching digit string, the TA900 will route outbound calls to the same trunk regardless of origin.
With SABR you can configure the device to use a specific outbound trunk based on the originating trunk or DID.
If both voice and fax calls are supposed use the same outbound trunk, then it should just work.
While I think it would be OK for the T01 trunk to handle everything if it will, SABR might be better a better option.
I have some questions after reading the SABR doc - Do I need ani lists for each service, one for the PRI users and one for the voice users?
And, for the ani list for the voice users - the voice user ani is the same as the voice user 'name', correct?
Thanks for the support. I have configured SABR fully for the PRI and voice users as outlined in the document that you indicated.
Unfortunately I am still waiting on confirmation that the fax line dials out from someone at the location.
But at least it does not appear to have caused any service issues on the PRI, and I see SABR being invoked on debug.
I will return to click the 'Question Answered' button once I get the final result of the fax line dailing out.
Here is the connection part of the debug: (T11 is Sip Server, T02 is the existing PRI)
05:03:30.356 SB.CALL 645 Idle Call sent from T11 to T02 (158152-fax001) 05:03:30.356 SB.CALL 645 State change >> Idle->Delivering
05:03:30.356 RTP.MANAGER Isdn(Group) 0/ - empty - RTP: Reserve resource
05:03:30.357 RTP.MANAGER Isdn(Group) 0/ - Dsp 0/1.1 - RTP: (null)
05:03:30.357 RTP.PROVIDER unknown - Dsp 0/1.1 - RTP: reserving already allocated RTP channel
05:03:30.357 TA.T11 01 TAInboundCall CallResp event accepted 05:03:30.357 TA.T11 01 State change >> TAInboundCall->TAConnectWaitIn (TAS_Calling) 05:03:30.358 TA.T02 01 State change >> TAIdle->TAOutGoing (TAS_Delivering)
05:03:30.358 TM.T02 01 tachg_Delivering
05:03:30.358 TM.T02 01 IsdnTmStateIdle->IsdnTmStateOutboundDeliver
05:03:30.358 TM.T02 01 IsdnTmStateOutboundDeliver::enter()
05:03:30.359 SB.CALL 645 Delivering Called the deliverResponse routine from Delivering 05:03:30.359 SB.CALL 645 Delivering DeliverResponse(accept) sent from T02 to T11