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    Completely agree, we might be putting an E1 card in the Cisco and
    front-ending it with a Patton yet...<br>
    <br>
    I have an SN4961 30 Channel license in stock.<br>
    <br>
    An update on my testing; At the moment, I have the Cisco set to
    rfc2833, and Wireshark (filter on 'rtpenvents') clearly shows the
    DTMF being originated from the Cisco.<br>
    <br>
    <br>
    I should also point out that it _looks_ like the PSTN&lt;--&gt;SIP
    gateway is doing DTMF in-band, which is passing transparently
    through FS, from 'external' to 'internal' i.e. DTMF works in-bound
    no matter which direction the call is placed.<br>
    <br>
    Next step: capture a trace of a 'good' vs 'bad' call between FS and
    the PSTN GW and verify which type of DTMF  the 'good' call is
    actually using.<br>
        This will also let me know if FS is converting RFC2833 to
    in-band correctly towards the PSTN.<br>
    <br>
    Also, as Yehavi pointed out, making the Cisco transcode might fix it
    (again, the trace between FS and PSTN GW might show something).<br>
    <br>
    Ahh, just looking at the Wiki I have check to make sure I have this
    correct;<br>
    <a
href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate</a><br>
    <br>
    Thanks everyone, <br>
    Mike<br>
    <br>
    On 24/05/2011 5:49 p.m., David Ponzone wrote:
    <blockquote cite="mid:8E8B36E9-30A4-4F2B-99D0-619A0432BCE6@ipeva.fr"
      type="cite">You should consider Patton SN4961.
      <div>I am no shareholder of Patton, but those boxes work so great
        and are quite cheaper than the competition, that it is hard to
        imagine why anyone would buy something else...</div>
      <div><br>
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                <div><font class="Apple-style-span" face="'Helvetica
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                      color="#1c00ff">David Ponzone  </font><font
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                          href="mailto:david.ponzone@ipeva.fr">david.ponzone@ipeva.fr</a></span></font></font></div>
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                                <div style="margin: 0px; text-align:
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                                    toutes les pièces jointes sont
                                    confidentiels et établis à
                                    l'intention exclusive de ses
                                    destinataires. Toute utilisation ou
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                                    interdite. Tout message électronique
                                    est susceptible d'altération. </i><b><i>IPeva</i></b><i> décline
                                    toute responsabilité au titre de ce
                                    message s'il a été altéré, déformé
                                    ou falsifié. Si vous n'êtes pas
                                    destinataire de ce message, merci de
                                    le détruire immédiatement et
                                    d'avertir l'expéditeur.</i></div>
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          <div>Le 24/05/2011 à 06:01, Yehavi Bourvine a écrit :</div>
          <br class="Apple-interchange-newline">
          <blockquote type="cite">
            <div dir="rtl">
              <div dir="ltr">Hello,</div>
              <div dir="ltr"> </div>
              <div dir="ltr">  I had for a long time a Cisco SIP-E1(with
                Q.sig) gateway which caused me much grief with DTMF; it
                works partially OK with INFO but not with RFC-2833. I
                filed a support call with Cisco, as its debug shows that
                it receives and detects the RFC2833 but nothing is sent
                to the E1 side.</div>
              <div dir="ltr"> </div>
              <div dir="ltr">The last conslusion of Cisco's support is
                that I have to enable transcoding but they coouldn't
                find how to enable it in my configuration. I now have an
                Audioodes Mediant-1000 on loan which does RFC-2833
                without any problems...</div>
              <div dir="ltr"> </div>
              <div dir="ltr">The only problem I've found so far with the
                AudioCodes is name passing via E1/Q.sig at the ringing
                phase (we use it to connect our Freeswitch to our Nortel
                PbX).</div>
              <div dir="ltr">
                 </div>
              <div dir="ltr">                        Regards, __Yehavi:</div>
              <div dir="ltr"> </div>
              <div dir="ltr"><br>
                <br>
              </div>
              <div dir="ltr" class="gmail_quote">
                <div>2011/5/24 Mike Taylor <span dir="ltr">&lt;<a
                      moz-do-not-send="true"
                      href="mailto:mtaylor@employees.org">mtaylor@employees.org</a>&gt;</span></div>
                <blockquote style="margin: 0px 0px 0px 0.8ex;
                  padding-left: 1ex; border-left: 1px solid rgb(204,
                  204, 204);" class="gmail_quote">
                  <div>
                    <div style="color: rgb(0, 0, 0); font-family:
                      Calibri,sans-serif; font-size: 14px; word-wrap:
                      break-word;">
                      <div>Good point, thanks,</div>
                      <div><br>
                      </div>
                      <div>But, RFC2833 I've found to be unreliable,
                        broken, patchy, scratchy…. </div>
                      <div><br>
                      </div>
                      <div>I've got it set to RFC2833 now, and it just
                        ain't working.</div>
                      <div><br>
                      </div>
                      <div>Will get more traces, trial and error until I
                        stumble across a good combo, or bug(s)…</div>
                      <div><br>
                      </div>
                      <div>Thanks again,</div>
                      <div>Mike</div>
                      <div><br>
                      </div>
                      <span>
                        <div style="border-width: 1pt medium medium;
                          border-style: solid none none; border-color:
                          rgb(181, 196, 223) currentcolor currentcolor;
                          padding: 3pt 0in 0in; text-align: left; color:
                          black; font-family: Calibri; font-size: 11pt;">
                          <span style="font-weight: bold;">From: </span>
                          Ognjen Seslija &lt;<a moz-do-not-send="true"
                            href="mailto:oseslija@gmail.com"
                            target="_blank">oseslija@gmail.com</a>&gt;<br>
                          <span style="font-weight: bold;">Reply-To: </span>
                          FreeSWITCH Users Help &lt;<a
                            moz-do-not-send="true"
                            href="mailto:freeswitch-users@lists.freeswitch.org"
                            target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt;<br>
                          <span style="font-weight: bold;">Date: </span>
                          Mon, 23 May 2011 20:46:09 +0200<br>
                          <span style="font-weight: bold;">To: </span>
                          FreeSWITCH Users Help &lt;<a
                            moz-do-not-send="true"
                            href="mailto:freeswitch-users@lists.freeswitch.org"
                            target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt;<br>
                          <span style="font-weight: bold;">Subject: </span>
                          Re: [Freeswitch-users] FS - DTMF and Cisco
                          CME/UCE<br>
                        </div>
                        <div>
                          <div class="h5">
                            <div><br>
                            </div>
                            Why use INFO when it's only method I know
                            Cisco won't speak (2833, kpml, sub/notify)?<br>
                            <br>
                            Here's the CME snippet for FS trunk I have.
                            FS sends 2833 tones to CME btw.<br>
                            <br>
                            dial-peer voice 100 voip<br>
                             description ** TO FS **<br>
                             destination-pattern .T<br>
                             b2bua<br>
                             session protocol sipv2<br>
                             session target sip-server<br>
                             session transport udp<br>
                             dtmf-relay rtp-nte<br>
                             codec g711alaw<br>
                             no vad<br>
                            <br>
                            <br>
                            <br>
                            <br>
                            <div class="gmail_quote">On Mon, May 23,
                              2011 at 4:07 AM, Mike Taylor <span
                                dir="ltr">&lt;<a moz-do-not-send="true"
                                  href="mailto:mtaylor@employees.org"
                                  target="_blank">mtaylor@employees.org</a>&gt;</span>
                              wrote:<br>
                              <blockquote style="margin: 0px 0px 0px
                                0.8ex; padding-left: 1ex; border-left:
                                1px solid rgb(204, 204, 204);"
                                class="gmail_quote">
                                <div style="color: rgb(0, 0, 0);
                                  font-family: Calibri,sans-serif;
                                  font-size: 14px; word-wrap:
                                  break-word;">
                                  <div>
                                    <div>
                                      <div>Hi,</div>
                                    </div>
                                  </div>
                                  <div><br>
                                  </div>
                                  <div>This is driving me nuts. </div>
                                  <div>I have a Cisco CME (a few
                                    actually) where I cannot get DTMF to
                                    work.</div>
                                  <div><br>
                                  </div>
                                  <div>Does someone happen to have the
                                    magic recipe of SIP/Dialplan
                                    profiles and Cisco config to make
                                    DTMF &lt;relay&gt; work?</div>
                                  <div><br>
                                  </div>
                                  <div>I spent several very very painful
                                    days trying to get RFC2833 to work,
                                    using FS to try and generate the
                                    tones itself. I ended up with all
                                    sorts of codec mismatches,
                                    delay,distortion and some really
                                    mangled DTMF tones…</div>
                                  <div><br>
                                  </div>
                                  <div>So, now I'm back to;</div>
                                  <div><br>
                                  </div>
                                  <div>PSTN----CiscoGW----&lt;external&gt;----FS----&lt;internal&gt;----&lt;CUST
                                    PABX&gt;---Phone</div>
                                  <div><br>
                                  </div>
                                  <div>Where &lt;CUST PABX&gt; is 3 CMEs
                                    that won't talk DTMF</div>
                                  <div>30+ other vendor's gateway's that
                                    WILL do DTMF…</div>
                                  <div><br>
                                  </div>
                                  <div>Context External and internal are
                                    both set to dtmf-type info (or
                                    dtmf_type info) </div>
                                  <div><br>
                                  </div>
                                  <div>DTMF works FROM the Cisco GW
                                    towards our network, doesn't </div>
                                  <div><br>
                                  </div>
                                  <div><br>
                                  </div>
                                  <div>So, does anyone have a magic
                                    solution?</div>
                                  <div> Because Cisco TAC say they see
                                    DTMF being sent (as rtp-nte), when I
                                    need INFO or in-band.</div>
                                  <div><br>
                                  </div>
                                  <div><br>
                                  </div>
                                  <div><br>
                                  </div>
                                  <div>Thanks,</div>
                                  <div>Mike</div>
                                </div>
                                <br>
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_______________________________________________
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<a class="moz-txt-link-abbreviated" href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a>
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<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
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