FreePBX Not able to send Fax

Hello

I am using FreePBX version 2.11.0.40 with asterisks version 11.13.1. . the issue I have is that I am unable to send fax using the FreePBX cup web portal. this was working before, but all of a sudden it just failed to send the fax. below is the error im getting from the logs.

== Spawn extension (outboundfax, s, 8) exited non-zero on 'SIP/etisalat-out-00000000'
    -- Executing [h@outboundfax:1] Set("SIP/etisalat-out-00000000", "FAXOPT(ecm)=no") in new stack
    -- Executing [h@outboundfax:2] Set("SIP/etisalat-out-00000000", "FAXOPT(headerinfo)=from Fax") in new stack
    -- Executing [h@outboundfax:3] Set("SIP/etisalat-out-00000000", "FAXOPT(localstationid)=+xxxxxxxxx") in new stack
    -- Executing [h@outboundfax:4] Set("SIP/etisalat-out-00000000", "FAXOPT(maxrate)=9600") in new stack
    -- Executing [h@outboundfax:5] Set("SIP/etisalat-out-00000000", "FAXOPT(minrate)=9600") in new stack
    -- Executing [h@outboundfax:6] NoOp("SIP/etisalat-out-00000000", "FaxStatus : FAILED") in new stack
    -- Executing [h@outboundfax:7] NoOp("SIP/etisalat-out-00000000", "FaxStatusString : The call dropped prematurely") in new stack
    -- Executing [h@outboundfax:8] NoOp("SIP/etisalat-out-00000000", "FaxError : The call dropped prematurely") in new stack
    -- Executing [h@outboundfax:9] NoOp("SIP/etisalat-out-00000000", "RemoteStationID : ") in new stack
    -- Executing [h@outboundfax:10] NoOp("SIP/etisalat-out-00000000", "FaxPages : 0") in new stack
    -- Executing [h@outboundfax:11] NoOp("SIP/etisalat-out-00000000", "FaxBitRate : 14400") in new stack
    -- Executing [h@outboundfax:12] NoOp("SIP/etisalat-out-00000000", "FaxResolution : 0x0") in new stack

The Telephone lines are delivered through a PRI line by the Telephone provider. I have tried multiple fax numbers and all of them fail. nothing changed on the PBX side.

The callee hung up prematurely, as it says in the messages:

this is happening to all the fax numbers I try, its not only 1 number, its multiple fax numbers I try.

That doesn’t change the fact that the log you provided shows that Asterisk thinks that the only thing that went wrong was the other party ending the call.

I have managed to get it to work. I have disabled T38 passthrough in “asterisk SIP Settings” in the freepbx GUI. Hope this helps somebody out there.

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