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    I suspect I followed a guide much like the one you have used
    including information found on voip-info - sorry, I can't seem to
    find any bookmarks of relevant information.<br>
    <br>
    I spent an enormous amount of time getting it working and working
    very well, the real issue was getting T.38 working - I applied a
    patch to Asterisk version 1.8 to get the T.38 gateway functionality.<br>
    <br>
    I would have started off my testing by confirming communications
    between two IAX modems, I presume you are using HylaFAX too.<br>
    <br>
    Once the communications between the two IAX modems was working I
    progressed with testing sending and receiving faxes using G711A
    through my VoIP service and a modem attached to a PSTN service,
    suffice to say T.38 functionality was the key to getting reliable
    faxes working through VoIP at least when traversing the Internet,
    fortunately my VoIP provider facilitates T.38.<br>
    <br>
    Using an SPA8800 on my network I tested sending and receiving faxes
    with a modem attached to the SPA8800, it worked in G711A and T.38.<br>
    <br>
    I progressed to Asterisk 11 where the T.38 gateway functionality is
    better along with other improvements.<br>
    <br>
    What is the output on your system for:<br>
    <br>
        fax show version<br>
    <br>
    <br>
    Cheers,<br>
    <br>
    Larry.<br>
    <br>
    <div class="moz-cite-prefix">On 15/11/2016 8:09 PM, tux john wrote:<br>
    </div>
    <blockquote
cite="mid:trinity-a1663328-acd3-45ba-ac15-7241cc84eb11-1479211745735@msvc-mesg-gmxus002"
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      <div class="mail_android_message" style="line-height: 1; padding:
        0.5em">Hi. Since I am messing a lot with it without seeing the
        end of, may I ask if there is any solid guide for that please?<br>
      </div>
      <div class="mail_android_quote" style="line-height: 1; padding:
        0.3em">On 13/11/2016, 07:42 Larry Moore
        <a class="moz-txt-link-rfc2396E" href="mailto:lmoore@starwon.com.au"><lmoore@starwon.com.au></a> wrote:
        <blockquote class="gmail_quote" style="margin: 0.8ex 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;"> Some additional information which may
          help you with your installation.<br>
          <br>
          I have 4 IAX Modems named iaxmodem0 - iaxmodem3. I use
          iaxmodem3 for outbound fax transmissions.<br>
          <br>
          I created a queue for the other 3 modems, here is my entry in
          queues.conf:<br>
          <blockquote> [hylafax-iax]<br>
            strategy=linear<br>
            ringinuse=yes<br>
            autopause=no<br>
            retry=4<br>
            timeout=5<br>
            timeoutpriority=conf<br>
            reportholdtime=no<br>
            joinempty=strict<br>
            leavewhenempty=strict<br>
            musicclass=silence<br>
            <br>
            member => IAX2/iaxmodem2<br>
            member => IAX2/iaxmodem1<br>
            member => IAX2/iaxmodem0<br>
            <br>
          </blockquote>
          In case you are wondering about the 'musicclass' I have used,
          here is the section from musiconhold.conf, the actual location
          of the files may be elsewhere on your system:<br>
          <blockquote> [silence]<br>
            mode=files<br>
            directory=/usr/local/share/asterisk/silence<br>
            ; ls /usr/local/share/asterisk/silence<br>
            ; 10.gsm<br>
            ;<br>
            ; The file 10.gsm came from
            /usr/local/share/asterisk/sounds/en/silence<br>
            <br>
          </blockquote>
          I changed 'callbackextension' in my sip.conf for the trunk so
          that it would go directly to the 'fax' extension in the
          dialplan i.e. 'callbackextension=fax'.<br>
          <br>
          I've included the console output when an incoming fax is
          received:<br>
          <blockquote>   == Using SIP RTP TOS bits 184<br>
                -- Executing [fax@from-itsp:1] NoOp("SIP/itsp-00000044",
            "Fax Detected 2016-11-13 12:33:40 +0800") in new stack<br>
                -- Executing [fax@from-itsp:2]
            GotoIf("SIP/itsp-00000044", "0?3:8") in new stack<br>
                -- Goto (from-itsp,fax,8)<br>
                -- Executing [fax@from-itsp:8] NoOp("SIP/itsp-00000044",
            "Finish if_from-itsp_237") in new stack<br>
                -- Executing [fax@from-itsp:9]
            GotoIf("SIP/itsp-00000044", "0?10:13") in new stack<br>
                -- Goto (from-itsp,fax,13)<br>
                -- Executing [fax@from-itsp:13]
            NoOp("SIP/itsp-00000044", "Finish if_from-itsp_238") in new
            stack<br>
                -- Executing [fax@from-itsp:14] Set("SIP/itsp-00000044",
            "FAXOPT(gateway)=yes") in new stack<br>
                -- Executing [fax@from-itsp:15]
            Queue("SIP/itsp-00000044", "hylafax-iax,dRt,,,15") in new
            stack<br>
                -- Started music on hold, class 'silence', on
            SIP/itsp-00000044<br>
                -- Call accepted by 127.0.0.1 (format alaw)<br>
                -- Format for call is (alaw)<br>
                -- IAX2/iaxmodem2-3086 is ringing<br>
                -- Stopped music on hold on SIP/itsp-00000044<br>
                -- IAX2/iaxmodem2-3086 answered SIP/itsp-00000044<br>
                   > 0x89bac000 -- Probation passed - setting RTP
            source address to <ITSP IP Address>:18998<br>
              == Using UDPTL TOS bits 184<br>
                -- Executing [h@from-itsp:1] GotoIf("SIP/itsp-00000044",
            "0?2:3") in new stack<br>
                -- Goto (from-itsp,h,3)<br>
                -- Executing [h@from-itsp:3] NoOp("SIP/itsp-00000044",
            "Finish if_from-itsp_239") in new stack<br>
                -- Executing [h@from-itsp:4] NoOp("SIP/itsp-00000044",
            "Call/Fax Ended 2016-11-13 12:36:41 +0800") in new stack<br>
                -- Hungup 'IAX2/iaxmodem2-3086'<br>
              == Spawn extension (from-itsp, fax, 15) exited non-zero on
            'SIP/itsp-00000044'<br>
            <br>
          </blockquote>
          I'm sure you've already checked and confirmed you have 'alaw'
          and 'ulaw' codecs permitted in your IAX Modems, iax.conf and
          sip.conf configurations<br>
          <br>
          To test your configuration you could set it up your
          environment so that you send an outgoing fax to yourself i.e.
          your dial your number at the VoIP provider, this assumes when
          you dial your VoIP number a connection is made back to you,
          you can then troubleshoot the communication.<br>
          <br>
          This is how I performed the majority of my tests.<br>
          <br>
          Not sure why you haven't explored the option of terminating a
          fax call in Asterisk, you will need some scripts to convert
          the received image to a PDF which is then e-mailed. An offer
          was made to you to provide scripts, if you set this up when
          your iaxmodem's aren't working a fallback will be for Asterisk
          to accept the call as it falls through, one thing you should
          know, if you use the T.38 Gateway in your dialplan you will
          need to disabled it prior to Asterisk terminating the call. I
          use extensions.ael so here is an example, I've included the
          macro I use to receive a fax in Asterisk:<br>
          <blockquote> context from-itsp {<br>
            <br>
                    s => {<br>
                            NoOp(Call Received ${STRFTIME(,,%F %T %z)});<br>
                            Set(CHANNEL(language)=en_AU);<br>
                            Set(DIALTIMEOUT=30);<br>
                            Progress();<br>
                            NoOp(Call Received from ${CALLERID(name)},
            Tel: ${CALLERID(num)});<br>
                    .<br>
                    . other conditions checked and extensions dialled<br>
                    .<br>
                    };<br>
            <br>
                    fax => {<br>
                            NoOp(Fax Detected ${STRFTIME(,,%F %T %z)});<br>
                            Set(FAXOPT(gateway)=yes);<br>
                            Queue(hylafax-iax,dRt,,,15);<br>
            <br>
                            Set(FAXOPT(gateway)=no);<br>
                           
            &fax-receive(<TSID>,<Header>,FaxMaster,lmoore);<br>
                            Hangup();<br>
                    };<br>
                           <br>
                    h => {<br>
                            if ( "X${FAXRXFILE}" != "X" )<br>
                            {<br>
                                    &email_rxfax();<br>
                            }<br>
                            NoOp(Call/Fax Ended ${STRFTIME(,,%F %T
            %z)});<br>
                    };<br>
            };<br>
            <br>
            macro fax-receive( fax-number, header-info, sender,
            recipient ) {<br>
            /*<br>
                    ${ARG1} is Receiving Station Fax Number<br>
                    ${ARG2} is Fax Header Information<br>
                    ${ARG3} is Fax Sender E-mail Address<br>
                    ${ARG4} is Fax Recipient E-mail Address<br>
            */<br>
                    NoOp(**** FAX RECEIVE ****);<br>
                    Set(FAXOPT(localstationid)=${LOCAL(fax-number)});<br>
                    Set(FAXOPT(headerinfo)=${LOCAL(header-info)});<br>
                    Set(FROMADDR=${LOCAL(sender)});<br>
                    Set(TOADDR=${LOCAL(recipient)});<br>
                    NoOp(**** SETTING FAXOPT ****);<br>
                    NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)});<br>
                    NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)});<br>
                    NoOp(FAXOPT(localstationid) :
            ${FAXOPT(localstationid)});<br>
                    Set(RXSTART=${EPOCH});<br>
                    Set(FAXRXPATH=/var/spool/asterisk/fax/received);<br>
                    Set(FAXRXFILE=fax-${CALLERID(number)}-${UNIQUEID});<br>
                    NoOp(**** RECEIVING FAX : ${FAXRXFILE} ****);<br>
                    ReceiveFAX(${FAXRXPATH}/${FAXRXFILE}.tif,f);<br>
                    NoOp(**** Subroutine Return ****);<br>
                    return;<br>
                    };<br>
          </blockquote>
          Cheers,<br>
          <br>
          Larry.<br>
          <br>
          <br>
          <div class="moz-cite-prefix"> On 13/11/2016 8:07 AM, Larry
            Moore wrote:<br>
          </div>
          <blockquote> Is your network/firewall configuration permitting
            the ports for UDPTL, runn the command:  udptl show config<br>
            <br>
            <blockquote> UDPTL Global options<br>
              --------------------<br>
              udptlstart:      4000<br>
              udptlend:        4999<br>
              udptlfecentries: 3<br>
              udptlfecspan:    3<br>
              use_even_ports:  No<br>
              udptlchecksums: Yes<br>
                <br>
            </blockquote>
            In your sip configuration for your 'mytrunk' peer have you
            set applicable options e.g.:<br>
            <blockquote> t38pt_udptl=yes,redundancy,maxdatagram=400<br>
            </blockquote>
            In your extensions.conf you could and probably should set
            the following option prior to dialing the IAX channel, this
            is to enable the T.38 gateway feature of Asterisk 11:<br>
            <blockquote> Set(FAXOPT(gateway)=yes)<br>
            </blockquote>
            I have it working in my installation however I have incoming
            voice calls too hence I use 'faxdetect' to direct the call
            to the 'fax' extension.<br>
            <br>
            Cheers,<br>
            <br>
            Larry.<br>
            <br>
            <div class="moz-cite-prefix"> On 12/11/2016 5:24 AM, tux
              john wrote:<br>
            </div>
            <blockquote>
              <div style="font-family: Verdana;font-size: 12.0px;">
                <div> hi. i am using asterisk 11.24.1 in my raspberry. i
                  do have a sip trunk with a provider with g711a. I am
                  trying to setup a fax server by following the guide in<a
                    moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://the-asterisk-book.com/1.6/faxserver.html" target="_blank">
                    <a class="moz-txt-link-freetext" href="http://the-asterisk-book.com/1.6/faxserver.html">http://the-asterisk-book.com/1.6/faxserver.html</a></a>.
                </div>
                <div>   </div>
                <div> i do live in Greece and the number is
                  00302112152130 </div>
                <div> the problem is that i am getting the following
                  error and i am stuck: </div>
                <div>   </div>
                <div>
                  <div>   == Using SIP RTP TOS bits 184<br>
                      == Using SIP RTP CoS mark 5<br>
                        -- Executing [00302112152130@fax-in:1]
                    Dial("SIP/mytrunk-00000001", "IAX2/iaxmodem") in new
                    stack<br>
                        -- Called IAX2/iaxmodem<br>
                        -- Hungup 'IAX2/iaxmodem-3818'<br>
                      == Everyone is busy/congested at this time
                    (1:0/0/1)<br>
                        -- Auto fallthrough, channel
                    'SIP/mytrunk-00000001' status is 'CHANUNAVAIL'<br>
                    RasPBX*CLI> </div>
                  <div>   </div>
                  <div>   </div>
                  <div> the extensions.conf has </div>
                  <div>
                    <div> [fax-in]<br>
                      exten => 00302112152130,1,Dial(IAX2/iaxmodem) </div>
                    <div>   </div>
                  </div>
                  <div>   </div>
                  <div> any ideas, please? </div>
                </div>
              </div>
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          <br>
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          the new Asterisk community forum at:<a moz-do-not-send="true"
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            https://community.asterisk.org/</a> New to Asterisk? Start
          here:<a moz-do-not-send="true"
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            http://lists.digium.com/mailman/listinfo/asterisk-users</a>
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