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    <div class="moz-cite-prefix">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.<br>
      <br>
      <pre class="moz-signature" cols="72">-- 
Yossi Neiman

On 04/07/2015 04:57 AM, Steven Ayre wrote:
</pre>
    </div>
    <blockquote
cite="mid:CAFiqYunOGTCj6emP7u5v=y2eCLr1_YLDa4FzELgzebdyTJ7Xyw@mail.gmail.com"
      type="cite">
      <div dir="ltr">That's not necessarily the case - GSM is not the
        only codec used by mobile networks, there'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 moz-do-not-send="true"
              href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc 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 class="HOEnZb">
              <div class="h5">
                <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
                        moz-do-not-send="true"
                        href="mailto:rtreleaven@bunnykick.ca"
                        target="_blank">rtreleaven@bunnykick.ca</a>&gt;</span>
                    wrote:<br>
                    <blockquote class="gmail_quote" style="margin:0 0 0
                      .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      <div dir="ltr">for the freeswitch side
                        <div>&lt;action application="set"
                          data="absolute_codec_string=L16@16000h"/&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 moz-do-not-send="true"
href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.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 dir="ltr">
                                  <div>
                                    <div>
                                      <div>thank you for reply<br>
                                      </div>
                                      it is freeswitch.conf<br>
                                       &lt;X-PRE-PROCESS cmd="set"
                                      data="sound_prefix=$${sounds_dir}/en/us/callie"/&gt;<br>
                                      &lt;X-PRE-PROCESS cmd="set"
                                      data="default_country=US"/&gt;<br>
                                      &lt;X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/&gt;<br>
                                      <br>
                                       &lt;profile
                                      name="freeswitch-sip"&gt;<br>
                                                &lt;gateways&gt;<br>
                                                  &lt;gateway
                                      name="asterisk-local"&gt;<br>
                                                    &lt;param
                                      name="proxy" value="<a
                                        moz-do-not-send="true"
                                        href="http://127.0.0.1:5060"
                                        target="_blank">127.0.0.1:5060</a>"/&gt;<br>
                                                    &lt;param
                                      name="retry-seconds"
                                      value="30"/&gt;<br>
                                                    &lt;param
                                      name="caller-id-in-from"
                                      value="true"/&gt;<br>
                                                  &lt;/gateway&gt;<br>
                                                &lt;/gateways&gt;<br>
                                                <br>
                                                &lt;domains&gt;<br>
                                                  &lt;domain name="all"
                                      alias="true" parse="false"/&gt; <br>
                                                &lt;/domains&gt;<br>
                                                <br>
                                                &lt;settings&gt;<br>
                                                  &lt;param name="debug"
                                      value="1"/&gt;<br>
                                                  &lt;param
                                      name="sip-trace" value="no"/&gt;<br>
                                                  &lt;param
                                      name="log-auth-failures"
                                      value="false"/&gt;<br>
                                                  &lt;param
                                      name="forward-unsolicited-mwi-notify"
                                      value="false"/&gt;<br>
                                                  &lt;param
                                      name="context"
                                      value="asterisk"/&gt;<br>
                                                  &lt;param
                                      name="rfc2833-pt" value="101"/&gt;<br>
                                                  &lt;param
                                      name="sip-port" value="5050"/&gt;<br>
                                                  &lt;param
                                      name="dialplan" value="XML"/&gt;<br>
                                                  &lt;param
                                      name="dtmf-type" value="info"/&gt;<br>
                                                  &lt;param
                                      name="inbound-codec-prefs"
                                      value="$${global_codec_prefs}"/&gt;<br>
                                                  &lt;param
                                      name="outbound-codec-prefs"
                                      value="$${global_codec_prefs}"/&gt;<br>
                                                  &lt;param
                                      name="use-rtp-timer"
                                      value="true"/&gt;<br>
                                                  &lt;param
                                      name="rtp-timer-name"
                                      value="soft"/&gt;<br>
                                                  &lt;param
                                      name="rtp-timeout-sec"
                                      value="300"/&gt;<br>
                                                  &lt;param
                                      name="rtp-hold-timeout-sec"
                                      value="1800"/&gt;<br>
                                                  &lt;param name="vad"
                                      value="none"/&gt; <br>
                                                  &lt;param
                                      name="rtp-ip"
                                      value="127.0.0.1"/&gt;<br>
                                                  &lt;param
                                      name="sip-ip"
                                      value="127.0.0.1"/&gt;<br>
                                                  &lt;param
                                      name="ext-rtp-ip"
                                      value="127.0.0.1"/&gt;<br>
                                                  &lt;param
                                      name="ext-sip-ip"
                                      value="127.0.0.1"/&gt;<br>
                                                  &lt;param
                                      name="inbound-codec-negotiation"
                                      value="generous"/&gt;<br>
                                                  &lt;param name="tls"
                                      value="false"/&gt;<br>
                                                  &lt;param
                                      name="nonce-ttl" value="60"/&gt;<br>
                                                  &lt;param
                                      name="auth-calls"
                                      value="false"/&gt;<br>
                                                  &lt;param
                                      name="auth-all-packets"
                                      value="false"/&gt;<br>
                                                  &lt;param
                                      name="challenge-realm"
                                      value="auto_from"/&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
                                            moz-do-not-send="true"
                                            href="mailto:rtreleaven@bunnykick.ca"
                                            target="_blank">rtreleaven@bunnykick.ca</a>&gt;</span>
                                        wrote:<br>
                                        <blockquote class="gmail_quote"
                                          style="margin:0 0 0
                                          .8ex;border-left:1px #ccc
                                          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
                                                      moz-do-not-send="true"
href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.com</a>&gt;</span>
                                                  wrote:<br>
                                                </div>
                                              </div>
                                              <blockquote
                                                class="gmail_quote"
                                                style="margin:0 0 0
                                                .8ex;border-left:1px
                                                #ccc
                                                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>
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        <br>
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      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
<a class="moz-txt-link-abbreviated" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a>
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