<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:10pt"><div class="" style=""><span class="" style="">Hi,</span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div style="background-color: transparent;" class=""><span class="" style="">i set faxopt(gateway)=yes and t38pt_udptl = yes,redundancy,maxdatagram=400 in sip.conf for the peer. <br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style=""><br class=""
 style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="">But s</span><span class="" style="font-size: 10pt;">till asterisk is sending </span><span class="" style="font-size: 10pt;">SIP 200 OK message with G711 and G729 codec in SDP as reply to T38 invite.</span></div><div style="color: rgb(0, 0, 0); font-size: 10pt; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="font-size: 10pt;"><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class=""
 style="font-size: 10pt;"><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="font-size: 10pt;">Everything will be fine if  itis  possbile to send SIP 488 when asterisk  receives T38 invite.  that is why i tried </span><span style="font-size: 10pt;" class=""> </span><span style="font-size: 10pt;" class="">t38pt_udptl = no</span></div><div style="color: rgb(0, 0, 0); font-size: 10pt; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span style="font-size: 10pt;" class=""><br></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande',
 sans-serif; font-style: normal; background-color: transparent;" class=""><span style="font-size: 10pt;" class=""><br></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span style="font-size: 10pt;" class=""><br></span></div><div style="color: rgb(0, 0, 0); font-size: 10pt; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="font-size: 10pt;">Best Regards</span></div><div style="color: rgb(0, 0, 0); font-size: 10pt; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="font-size: 10pt;"><br></span></div><div style="color: rgb(0, 0, 0); font-size:
 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class=""
 style=""><br class="" style=""></span></div><div class="" style=""><br class="" style=""></div>  <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 10pt;" class=""> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 12pt;" class=""> <div dir="ltr" class="" style=""> <hr size="1" class="" style="">  <font size="2" face="Arial" class="" style=""> <b class="" style=""><span style="font-weight:bold;" class="">From:</span></b> Gregory Massel <greg@csurf.co.za><br class="" style=""> <b class="" style=""><span style="font-weight: bold;" class="">To:</span></b> asterisk-ss7@lists.digium.com <br class="" style=""> <b class="" style=""><span style="font-weight: bold;" class="">Sent:</span></b> Tuesday, August 26, 2014 9:35 AM<br class="" style=""> <b class="" style=""><span style="font-weight: bold;" class="">Subject:</span></b> Re:
 [asterisk-ss7] T38 fax issue<br class="" style=""> </font> </div> <div class="" style=""><br class="" style=""><div id="yiv9250629107" class="" style=""><div class="" style="">
    See <a rel="nofollow" shape="rect" class="" target="_blank" href="https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway" style="">https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway</a><br clear="none" class="" style="">
    <br clear="none" class="" style="">
    Particularly the line: <br clear="none" class="" style="">
    <pre class="" style="">exten => 1,n,Set(FAXOPT(gateway)=yes)

which you need on the SIP side of the DAHDI-SIP bridge
</pre>
    <div class="" id="yiv9250629107yqt43518" style=""><div class="" style="">On 2014/08/26 08:01 AM, Huseyin Kaya
      wrote:<br clear="none" class="" style="">
    </div>
    <blockquote type="cite" class="" style="">
      <div style="color:#000;background-color:#fff;font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:10pt;" class="">
        <div class="" style=""><span class="" style="">Hello</span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style=""><br clear="none" class="" style="">
          </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="">Does anyone has
            a working libb7 server that able to handle T38 faxing.</span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style=""><br clear="none" class="" style="">
          </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="">I am working on
            this for the last 10 days. But still no solution. </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style=""><br clear="none" class="" style="">
          </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="">I am receiving
            the call from sip and sending to telco with ss7. </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style=""><br clear="none" class="" style="">
          </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="">At the
            beginning of the call everything is fine ( i mean codec
            negotiation) . Then the remote sip side detects the fax tone
            that remote end sends and  </span><span style="background-color:transparent;" class="">Then the sip remote
            side sends the t38 invite and asterisk sends SIP 200 OK
            message with G711 and G729 codec in SDP.</span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span style="background-color:transparent;" class=""><br clear="none" class="" style="">
          </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span style="background-color:transparent;" class="">My customer is saying that if i am sending SIP
            200 OK to his T38 invite .in SDP of SIP 200 OK there should
            be only T38.</span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span style="background-color:transparent;" class=""> </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida           Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style=""><br clear="none" class="" style="">
          </span></div>
        <div class="" style=""><span class="" style="font-size:13px;">Also
            disabling t38 with t38pt_udptl=no didnt change anything.
             Still asterisk is sending </span><span class="" style="font-size:10pt;">SIP 200 OK message with G711 and G729 codec
            in SDP as reply to T38 invite.</span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:10pt;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="font-size:10pt;"><br clear="none" class="" style="">
          </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="font-size:10pt;"><br clear="none" class="" style="">
          </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="font-size:10pt;">I will be glad if current libss7 users can advice
            me a way to find a solution on this issue</span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:10pt;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="font-size:10pt;"><br clear="none" class="" style="">
          </span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-style:normal;background-color:transparent;"><span class="" style="font-size:10pt;">Regards</span></div>
        <div class="" style="color:rgb(0, 0, 0);font-size:10pt;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-style:normal;background-color:transparent;"><br clear="none" class="" style="">
        </div>
        <div class="" style="color:rgb(0, 0, 0);font-size:13px;font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-style:normal;background-color:transparent;"> </div>
        <div class="" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:10pt;">
          <div class="" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:12pt;">
            <div class="" dir="ltr" style="">
              <hr class="" style="" size="1"> <font class="" style="" size="2" face="Arial"> <b class="" style=""><span class="" style="font-weight:bold;">From:</span></b>
                Huseyin Kaya <a rel="nofollow" shape="rect" class="" ymailto="mailto:huseyinkaya@yahoo.com" target="_blank" href="mailto:huseyinkaya@yahoo.com" style=""><huseyinkaya@yahoo.com></a><br clear="none" class="" style="">
                <b class="" style=""><span class="" style="font-weight:bold;">To:</span></b> Kaloyan Kovachev
                <a rel="nofollow" shape="rect" class="" ymailto="mailto:kkovachev@varna.net" target="_blank" href="mailto:kkovachev@varna.net" style=""><kkovachev@varna.net></a>;
                <a rel="nofollow" shape="rect" class="" ymailto="mailto:asterisk-ss7@lists.digium.com" target="_blank" href="mailto:asterisk-ss7@lists.digium.com" style="">"asterisk-ss7@lists.digium.com"</a>
                <a rel="nofollow" shape="rect" class="" ymailto="mailto:asterisk-ss7@lists.digium.com" target="_blank" href="mailto:asterisk-ss7@lists.digium.com" style=""><asterisk-ss7@lists.digium.com></a> <br clear="none" class="" style="">
                <b class="" style=""><span class="" style="font-weight:bold;">Sent:</span></b> Friday, August 22, 2014
                5:32 PM<br clear="none" class="" style="">
                <b class="" style=""><span class="" style="font-weight:bold;">Subject:</span></b> Re: [asterisk-ss7] T38
                fax issue<br clear="none" class="" style="">
              </font> </div>
            <div class="" style=""><br clear="none" class="" style="">
              <div class="" id="yiv9250629107" style="">
                <div class="" style="">
                  <table class="" style="" border="0" cellpadding="0" cellspacing="0"><tbody class="" style=""><tr class="" style=""><td colspan="1" rowspan="1" valign="top" class="" style="">
                          <div class="" dir="ltr" style=""><font class="" style="" size="2"><font class="" style="" size="2">Hello</font></font><br clear="none" class="" style="">
                          </div>
                          <div class="" dir="ltr" style=""><font class="" style="" size="2"><font class="" style="" size="2">Actually it will be
                                fine If I can send sip 488 not
                                acceptable here to the re invite of t38
                                .</font></font></div>
                          <div class="" dir="ltr" style=""><font class="" style="" size="2"><font class="" style="" size="2">But Asterisk is
                                sending sip 200 ok with voice codecs in
                                SDP.</font></font></div>
                          <div class="" dir="ltr" style=""><font class="" style="" size="2"><font class="" style="" size="2">So if i was able to
                                send sip 488 . The fax will contunie
                                with g711 </font></font></div>
                          <div class="" dir="ltr" style=""><font class="" style="" size="2"><font class="" style="" size="2">Setting
                                faxopt(gateway)=yes and t38pt_udptl =
                                yes,redundancy,maxdatagram=400 didnt
                                work. </font></font><br clear="none" class="" style="">
                            <font class="" style="" size="2"><font class="" style="" size="2">By setting
                                these asterisk should send sip 200 with
                                SDP T38 but it is still sending speech
                                codec(g11u,etc..) in SIP 200 sdp.</font></font></div>
                          <div class="" dir="ltr" style=""><font class="" style="" size="2"><font class="" style="" size="2">Also disabling t38
                                with t38pt_udptl=no didnt change
                                anything. </font></font><br clear="none" class="" style="">
                            <br clear="none" class="" style="">
                          </div>
                          <div class="" dir="ltr" style=""><font class="" style="" size="2"><font class="" style="" size="2">Regards</font></font><br clear="none" class="" style="">
                          </div>
                        </td></tr></tbody></table>
                  <div class="" id="yiv9250629107yqt73711" style="">
                    <div class="" id="yiv9250629107_origMsg_" style="">
                      <div class="" style=""> <br clear="none" class="" style="">
                        <div class="" style="">
                          <div class="" style="font-size:0.9em;">
                            <hr class="" style="" size="1"> <b class="" style=""> <span class="" style="font-weight:bold;">From:</span> </b> Kaloyan
                            Kovachev <a rel="nofollow" shape="rect" class="" ymailto="mailto:kkovachev@varna.net" target="_blank" href="mailto:kkovachev@varna.net" style=""><kkovachev@varna.net></a>; <br clear="none" class="" style="">
                            <b class="" style=""> <span class="" style="font-weight:bold;">To:</span>
                            </b> Huseyin Kaya
                            <a rel="nofollow" shape="rect" class="" ymailto="mailto:huseyinkaya@yahoo.com" target="_blank" href="mailto:huseyinkaya@yahoo.com" style=""><huseyinkaya@yahoo.com></a>;
                            <a rel="nofollow" shape="rect" class="" ymailto="mailto:asterisk-ss7@lists.digium.com" target="_blank" href="mailto:asterisk-ss7@lists.digium.com" style=""><asterisk-ss7@lists.digium.com></a>; <br clear="none" class="" style="">
                            <b class="" style=""> <span class="" style="font-weight:bold;">Subject:</span>
                            </b> Re: [asterisk-ss7] T38 fax issue <br clear="none" class="" style="">
                            <b class="" style=""> <span class="" style="font-weight:bold;">Sent:</span>
                            </b> Fri, Aug 22, 2014 1:52:44 PM <br clear="none" class="" style="">
                          </div>
                          <br clear="none" class="" style="">
                          <table class="" style="" border="0" cellpadding="0" cellspacing="0"><tbody class="" style=""><tr class="" style=""><td colspan="1" rowspan="1" valign="top" class="" style="">Hi,<br clear="none" class="" style="">
                                  in addition to FAXOPT(gateway) you may
                                  try to request transmission <br clear="none" class="" style="">
                                  medium from telco - see SS7_TMR or
                                  SS7_TMR_NUM, as it should be set on <br clear="none" class="" style="">
                                  the outgoing (dahdi) channel you need
                                  to set it on the SIP channel with <br clear="none" class="" style="">
                                  underscore (_SS7_TMR)<br clear="none" class="" style="">
                                  <br clear="none" class="" style="">
                                  The possible options are defined in
                                  libss7.h and SS7_TMR_3K1_AUDIO (or 3 <br clear="none" class="" style="">
                                  as num) works fine here. If you can
                                  detect the fax calls you may request <br clear="none" class="" style="">
                                  64K_UNRESTRICTED data for them<br clear="none" class="" style="">
                                  <div class="" id="yiv9250629107yqtfd89673" style=""><br clear="none" class="" style="">
                                    On 2014-08-22 15:47, Huseyin Kaya
                                    wrote:<br clear="none" class="" style="">
                                    <br clear="none" class="" style="">
                                    > Hello<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > I am using Sangoma A104DE with
                                    libss7 on Asterisk 11.5.0 and <br clear="none" class="" style="">
                                    > terminating calls to telco with
                                    ss7 . We are using patched version
                                    of <br clear="none" class="" style="">
                                    > Libss7 that have timers <br clear="none" class="" style="">
                                    > functionality.(<a rel="nofollow" shape="rect" class="" target="_blank" href="https://issues.asterisk.org/jira/browse/SS7-27" style="">https://issues.asterisk.org/jira/browse/SS7-27</a>)<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > The server is on production for
                                    the last 6 months and everything was
                                    <br clear="none" class="" style="">
                                    > fine<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > My interconnection with telco
                                    is only one way. >From sip to ss7 .<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > Everything is fine except one
                                    thing.<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > One of my customers asked for
                                    t38 faxing. then i found myself in <br clear="none" class="" style="">
                                    > trouble.<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > I get several sip traces and
                                    found the problem at the end .When i
                                    <br clear="none" class="" style="">
                                    > receive an invite T38 ,
                                    asterisk is sending 200 OK message
                                    but in SDP <br clear="none" class="" style="">
                                    > it is sending g711u,g711a,g729
                                    as codecs. So after this point <br clear="none" class="" style="">
                                    > everything is messed up.<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > I tried to set t38pt_udptl=no
                                    in sip.conf , but still asterisk is
                                    <br clear="none" class="" style="">
                                    > sending sip 200 ok with sdp
                                    g711...<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > ı tried to set t38pt_udptl =
                                    yes,redundancy,maxdatagram=400 and <br clear="none" class="" style="">
                                    > setvar=FAXOPT(gateway)=yes,20
                                    in sip.conf but still i couldn't
                                    manage <br clear="none" class="" style="">
                                    > to receive fax<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > So basically what i need to
                                    send fax from sip to telco but could
                                    not <br clear="none" class="" style="">
                                    > manage to do till now.<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > Everything except fax is
                                    working like a charm.<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > Is anyone succeed to handle fax
                                    with libss7.<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > I will be glad if an expert can
                                    show me a way to achive this.<br clear="none" class="" style="">
                                    > <br clear="none" class="" style="">
                                    > Best Regards<br clear="none" class="" style="">
                                  </div>
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              </div>
              <br clear="none" class="" style="">
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      </div>
      <br clear="none" class="" style="">
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    </blockquote></div>
    <br clear="none" class="" style="">
  </div></div><br class="" style=""><div class="" id="yqt77920" style="">-- <br clear="none" class="" style="">_____________________________________________________________________<br clear="none" class="" style="">-- Bandwidth and Colocation Provided by <a shape="rect" href="http://www.api-digital.com/" target="_blank" class="" style="">http://www.api-digital.com </a>--<br clear="none" class="" style=""><br clear="none" class="" style="">asterisk-ss7 mailing list<br clear="none" class="" style="">To UNSUBSCRIBE or update options visit:<br clear="none" class="" style="">   <a shape="rect" href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank" class="" style="">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a></div><br class="" style=""><br class="" style=""></div> </div> </div>  </div></body></html>