nice to know it worked! old school? there are billions of POTS lines still using it although they&#39;re feed by digital exchanges:-)<br><br><div class="gmail_quote">On Mon, Apr 16, 2012 at 12:46 AM, Brian Foster <span dir="ltr">&lt;<a href="mailto:bdfoster@endigotech.com">bdfoster@endigotech.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>It is what your telephone company&#39;s equipment does, not your mobile phone provider. </p><div class="HOEnZb"><div class="h5">

<div class="gmail_quote">On Apr 15, 2012 12:39 PM, &quot;ocset&quot; &lt;<a href="mailto:ocset@the800group.com" target="_blank">ocset@the800group.com</a>&gt; wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


  
    
  
  <div bgcolor="#FFFFFF" text="#000000">
    Hi Nandy<br>
    <br>
    Thanks for you help. <br>
    <br>
    Just to answer you question first. It makes not difference who hangs
    up first, the calls never terminate.<br>
    <br>
    I have enabled &quot;Polarity Reversal&quot; on the GXW4104 and that has
    solved the problem. What I don&#39;t get is why this would work. I
    thought Polarity Reversal is old school and how would my mobile
    phone be doing this or is this being done by the mobile phone
    provider/carrier?<br>
    <br>
    I now have &quot;Enable Current Disconnect&quot;, &quot;Enable Tone Disconnect&quot; and
    &quot;Enable Polarity Reversal&quot; set for good measure even though the
    &quot;Reverse Polarity&quot; was the one that seemed to fix the problem.<br>
    <br>
    Regards<br>
    O.<br>
    <br>
    On 04/15/2012 09:01 AM, Nandy Dagondon wrote:
    <blockquote type="cite">GSM device or not, just determine what line disconnect
      signalling is used in each port. the GWX gateway must be set to
      detect this. otherwise, there&#39;s no way FS can hangup the call.
      this signal must be present when using the SIP gateway.<br>
      <br>
      also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2
      hangs up first.<br>
      <br>
      <div class="gmail_quote">On Sun, Apr 15, 2012 at 12:48 AM, ocset <span dir="ltr">&lt;<a href="mailto:ocset@the800group.com" target="_blank">ocset@the800group.com</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
          <div bgcolor="#FFFFFF" text="#000000"> Thanks Nandy<br>
            <br>
            I am not using the GSM device in this test and have tried
            changing the disconnect signalling without any luck. When I
            hang up the phones, nothing is recorder in the Freeswitch
            log (log level 7). Is there a way to trap the disconnect
            signal by writing a &quot;lua&quot; script or would that info already
            be written to the log if it was being received?.<br>
            <br>
            What I don&#39;t understand is that when I use the SIP gateway
            for the outgoing leg of the call, then the disconnect signal
            from the two phones is received. But when I use the PSTN
            line as the outgoing leg, the disconnect signal is not
            received from either phone?<br>
            <br>
            Anyone got any suggestions how I would debug this?<br>
            <br>
            Thanks again for your help.<span><font color="#888888"><br>
                O</font></span>
            <div>
              <div><br>
                <br>
                <br>
                On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
                <blockquote type="cite">check the line supervision
                  signalling of your GSM and the GXW4104. there are
                  several disconnect signalling used: current
                  disconnect, polarity reversal or busy tones. your FXO
                  port must be set accordingly. <br>
                  <br>
                  <div class="gmail_quote"> On Fri, Apr 13, 2012 at 3:09
                    PM, ocset <span dir="ltr">&lt;<a href="mailto:ocset@the800group.com" target="_blank">ocset@the800group.com</a>&gt;</span>
                    wrote:<br>
                    <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      Hi<br>
                      <br>
                      I have successfully managed to get FreeSwitch to
                      bridge an incoming call<br>
                      on a PSTN line to an external mobile phone using a
                      second PSTN line with<br>
                      the help of a GXW4104 FSO gateway.<br>
                      <br>
                      I am now having a serious issue where the call
                      never gets terminated<br>
                      when the parties hang up. The only way to end the
                      call is to either<br>
                      shut-down FreeSwitch or turn off the GXW4104. If I
                      bridge the call to a<br>
                      SIP gateway (PennyTel) instead of the second PSTN
                      line, then it works as<br>
                      expected when both parties hang up.<br>
                      <br>
                      Here is what I have tested<br>
                      <br>
                      1. Call -&gt; PSTN line1 (GXW4140) -&gt; PennyTel
                      SIP gateway -&gt; external<br>
                      mobile phone -- call end when both parties hang
                      up.<br>
                      2. Call -&gt; PSTN line1 (GXW4104) -&gt; ¬†PSTN
                      line2 (GXW4104) -&gt; external<br>
                      mobile phone -- call never end, even when both
                      parties hang up.<br>
                      <br>
                      Is there a parameter I should be setting to help
                      solve this issue or<br>
                      some code that I should implement to test for this
                      situation?<br>
                      <br>
                      I have also noticed that with test scenario 1
                      (PSTN line + SIP Gateway),<br>
                      if the person being called does not end the call
                      on their side, then<br>
                      that leg of the bridge remains active.<br>
                      <br>
                      ps. The second PSTN line is not a real copper
                      line. It is a PennyTel<br>
                      VOIP line that is managed by the modem and
                      connected to the GXW4104 via<br>
                      phone cable. So, as far as the GXW4104 is
                      concerned, it is just another<br>
                      POTS line. I don&#39;t have a second PSTN phone line
                      so can&#39;t test any other<br>
                      way.<br>
                      <br>
                      Thanks in advance<br>
                      O.<br>
                      <br>
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                    </blockquote>
                  </div>
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                </blockquote>
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    <br>
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