8 Replies Latest reply on Feb 24, 2020 12:08 PM by sharpy

    How to have FXS voice users (fax) use different SIP Server than the ISDN PRI port, or should we do NAT instead?

    sharpy New Member

      Hello,

       

      We currently have a TA908e that has an active ISDN PRI running, but we need to add fax lines currently on another ATA to it. We need to either:

      1. Add voice users for the FXS ports, but they have to use a different sip-server than the PRI - how to set that up?

      OR

      2. Enable the 2nd ethernet port on the TA and do NAT, and move the existing ATA to that port. But we don't want to affect the current operation of the TA with regards to the PRI. Is that possible?

       

      Please advise, thanks

        • Re: How to have FXS voice users (fax) use different SIP Server than the ISDN PRI port, or should we do NAT instead?
          jayh Hall_of_Fame

          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.

            • Re: How to have FXS voice users (fax) use different SIP Server than the ISDN PRI port, or should we do NAT instead?
              sharpy New Member

              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:

               

              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

                password "1234"

                forward-disconnect delay 250

                sip-identity 158152-fax001 T11 register auth-name "158152-fax001" password "1234VXh84321"

                codec-list SIP-Codec

               

              These are the only added lines to support the fxs users. They are registered on the sip server.

               

              However:

               

              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?

              • Re: How to have FXS voice users (fax) use different SIP Server than the ISDN PRI port, or should we do NAT instead?
                sharpy New Member

                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