<div dir="ltr"><div class="gmail_extra">Yes, GSM dongle can do AMR, but the I/O (with PC) in L16. So, the best codec, in this case, will be L16.<br></div><div class="gmail_extra"><br></div><div class="gmail_extra"><div class="gmail_quote">2015-04-07 19:59 GMT+03:00 Russell Treleaven <span dir="ltr">&lt;<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>&gt;</span>:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Can the GSM dongle do AMR?</div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Apr 7, 2015 at 12:42 PM, I put the Who? in Mishehu <span dir="ltr">&lt;<a href="mailto:mishehu@freeswitch.org" target="_blank">mishehu@freeswitch.org</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
  
    
  
  <div bgcolor="#FFFFFF" text="#000000">
    <div>I think that AMR is one of the codecs
      that does not allow for transcoding, at least with the in-tree
      module.  Might require something additional.<span><font color="#888888"><br>
      <br>
      <pre cols="72">-- 
Yossi Neiman

On 04/07/2015 04:57 AM, Steven Ayre wrote:
</pre>
    </font></span></div><div><div>
    <blockquote type="cite">
      <div dir="ltr">That&#39;s not necessarily the case - GSM is not the
        only codec used by mobile networks, there&#39;s others such as <span style="color:rgb(37,37,37);font-family:sans-serif;font-size:14px;line-height:22.3999996185303px">AMR</span></div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On 4 April 2015 at 19:11, Russell
          Treleaven <span dir="ltr">&lt;<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
            <div dir="ltr">Since one of the legs is GSM the best you can
              hope for is slightly worse than GSM because you are
              transcoding.
              <div><br>
              </div>
            </div>
            <div>
              <div>
                <div class="gmail_extra"><br>
                  <div class="gmail_quote">On Sat, Apr 4, 2015 at 2:01
                    PM, Russell Treleaven <span dir="ltr">&lt;<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>&gt;</span>
                    wrote:<br>
                    <blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
                      <div dir="ltr">for the freeswitch side
                        <div>&lt;action application=&quot;set&quot;
                          data=&quot;absolute_codec_string=L16@16000h&quot;/&gt;</div>
                        <div><br>
                        </div>
                        <div>for the asterisk side</div>
                        <div>you will have to figure that out yourself.</div>
                        <div><br>
                          <div><br>
                          </div>
                        </div>
                      </div>
                      <div>
                        <div>
                          <div class="gmail_extra"><br>
                            <div class="gmail_quote">On Sat, Apr 4, 2015
                              at 12:40 PM, Shabbir abbasi <span dir="ltr">&lt;<a href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.com</a>&gt;</span>
                              wrote:<br>
                              <blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
                                <div dir="ltr">
                                  <div>
                                    <div>
                                      <div>thank you for reply<br>
                                      </div>
                                      it is freeswitch.conf<br>
                                       &lt;X-PRE-PROCESS cmd=&quot;set&quot;
                                      data=&quot;sound_prefix=$${sounds_dir}/en/us/callie&quot;/&gt;<br>
                                      &lt;X-PRE-PROCESS cmd=&quot;set&quot;
                                      data=&quot;default_country=US&quot;/&gt;<br>
                                      &lt;X-PRE-PROCESS cmd=&quot;set&quot;
data=&quot;global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM&quot;/&gt;<br>
                                      <br>
                                       &lt;profile
                                      name=&quot;freeswitch-sip&quot;&gt;<br>
                                                &lt;gateways&gt;<br>
                                                  &lt;gateway
                                      name=&quot;asterisk-local&quot;&gt;<br>
                                                    &lt;param
                                      name=&quot;proxy&quot; value=&quot;<a href="http://127.0.0.1:5060" target="_blank">127.0.0.1:5060</a>&quot;/&gt;<br>
                                                    &lt;param
                                      name=&quot;retry-seconds&quot;
                                      value=&quot;30&quot;/&gt;<br>
                                                    &lt;param
                                      name=&quot;caller-id-in-from&quot;
                                      value=&quot;true&quot;/&gt;<br>
                                                  &lt;/gateway&gt;<br>
                                                &lt;/gateways&gt;<br>
                                                <br>
                                                &lt;domains&gt;<br>
                                                  &lt;domain name=&quot;all&quot;
                                      alias=&quot;true&quot; parse=&quot;false&quot;/&gt; <br>
                                                &lt;/domains&gt;<br>
                                                <br>
                                                &lt;settings&gt;<br>
                                                  &lt;param name=&quot;debug&quot;
                                      value=&quot;1&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;sip-trace&quot; value=&quot;no&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;log-auth-failures&quot;
                                      value=&quot;false&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;forward-unsolicited-mwi-notify&quot;
                                      value=&quot;false&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;context&quot;
                                      value=&quot;asterisk&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;rfc2833-pt&quot; value=&quot;101&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;sip-port&quot; value=&quot;5050&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;dialplan&quot; value=&quot;XML&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;dtmf-type&quot; value=&quot;info&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;inbound-codec-prefs&quot;
                                      value=&quot;$${global_codec_prefs}&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;outbound-codec-prefs&quot;
                                      value=&quot;$${global_codec_prefs}&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;use-rtp-timer&quot;
                                      value=&quot;true&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;rtp-timer-name&quot;
                                      value=&quot;soft&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;rtp-timeout-sec&quot;
                                      value=&quot;300&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;rtp-hold-timeout-sec&quot;
                                      value=&quot;1800&quot;/&gt;<br>
                                                  &lt;param name=&quot;vad&quot;
                                      value=&quot;none&quot;/&gt; <br>
                                                  &lt;param
                                      name=&quot;rtp-ip&quot;
                                      value=&quot;127.0.0.1&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;sip-ip&quot;
                                      value=&quot;127.0.0.1&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;ext-rtp-ip&quot;
                                      value=&quot;127.0.0.1&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;ext-sip-ip&quot;
                                      value=&quot;127.0.0.1&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;inbound-codec-negotiation&quot;
                                      value=&quot;generous&quot;/&gt;<br>
                                                  &lt;param name=&quot;tls&quot;
                                      value=&quot;false&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;nonce-ttl&quot; value=&quot;60&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;auth-calls&quot;
                                      value=&quot;false&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;auth-all-packets&quot;
                                      value=&quot;false&quot;/&gt;<br>
                                                  &lt;param
                                      name=&quot;challenge-realm&quot;
                                      value=&quot;auto_from&quot;/&gt;<br>
                                                &lt;/settings&gt;<br>
                                      <br>
                                      <br>
                                    </div>
                                    and  here is asterisk  sip.con<br>
                                    disallow=all<br>
                                    allow=ulaw<br>
                                    allow=alaw<br>
                                    <br>
                                  </div>
                                  what i need to change  ??<br>
                                </div>
                                <div>
                                  <div>
                                    <div class="gmail_extra"><br>
                                      <div class="gmail_quote">On Sat,
                                        Apr 4, 2015 at 9:13 PM, Russell
                                        Treleaven <span dir="ltr">&lt;<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>&gt;</span>
                                        wrote:<br>
                                        <blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
                                          <div dir="ltr">
                                            <div>probably the best you
                                              can do is limit the number
                                              of transcodings and/or
                                              resamplings</div>
                                            <div><br>
                                            </div>
                                            <div>
                                              <div>skype uses silk</div>
                                              <div><span style="white-space:pre-wrap">
                                                </span>freeswitch core
                                                uses L16</div>
                                              <div><span style="white-space:pre-wrap">
                                                </span>sip session uses
                                                &lt;your choice&gt;</div>
                                              <div><span style="white-space:pre-wrap">
                                                </span>asterisk core
                                                uses ?</div>
                                              <div><span style="white-space:pre-wrap">
                                                </span>audio is
                                                presented to the dongle
                                                as ?</div>
                                              <div><span style="white-space:pre-wrap">
                                                </span>cellular network
                                                uses gsm</div>
                                              <div><br>
                                              </div>
                                              <div>if ? =  L16 then make
                                                the sip session use L16</div>
                                              <div>use the loopback
                                                interface so that MTU is
                                                not an issue</div>
                                            </div>
                                            <div><br>
                                            </div>
                                          </div>
                                          <div class="gmail_extra"><br>
                                            <div class="gmail_quote">
                                              <div>
                                                <div>On Fri, Apr 3, 2015
                                                  at 2:50 PM, Shabbir
                                                  abbasi <span dir="ltr">&lt;<a href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.com</a>&gt;</span>
                                                  wrote:<br>
                                                </div>
                                              </div>
                                              <blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
                                                <div>
                                                  <div>
                                                    <div dir="ltr">
                                                      <div>for this
                                                        setup<br>
                                                      </div>
                                                      skype   --&gt;   
                                                      freeswitch(mod_skypopen
                                                      --&gt; mod_sofia)
                                                      ---&gt;
                                                      asterisk(chan_sip
                                                      --&gt;
                                                      chan_dongle[Huawei
                                                      E169] )     on
                                                      same machine <br>
                                                      <br>
                                                      suggest best
                                                      codecs to use with
                                                      freeswitch  and
                                                      asterisk  for good
                                                      voice quality ??<br>
                                                    </div>
                                                    <br>
                                                  </div>
                                                </div>
                                                <span>_________________________________________________________________________<br>
                                                  Professional
                                                  FreeSWITCH Consulting
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                                                </span></blockquote>
                                            </div>
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                              </blockquote>
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                            <br>
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                        </div>
                      </div>
                    </blockquote>
                  </div>
                  <br>
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          </blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_________________________________________________________________________
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<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></pre>
    </blockquote>
    <br>
  </div></div></div>

<br>_________________________________________________________________________<br>
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<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div>
</div></div><br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div></div>