<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=$${sounds_dir}/en/us/callie&quot;/&gt;<br>&lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;default_country=US&quot;/&gt;<br>&lt;X-PRE-PROCESS cmd=&quot;set&quot; data=&quot;global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM&quot;/&gt;<br><br> &lt;profile name=&quot;freeswitch-sip&quot;&gt;<br>          &lt;gateways&gt;<br>            &lt;gateway name=&quot;asterisk-local&quot;&gt;<br>              &lt;param name=&quot;proxy&quot; value=&quot;<a href="http://127.0.0.1:5060">127.0.0.1:5060</a>&quot;/&gt;<br>              &lt;param name=&quot;retry-seconds&quot; value=&quot;30&quot;/&gt;<br>              &lt;param name=&quot;caller-id-in-from&quot; value=&quot;true&quot;/&gt;<br>            &lt;/gateway&gt;<br>          &lt;/gateways&gt;<br>          <br>          &lt;domains&gt;<br>            &lt;domain name=&quot;all&quot; alias=&quot;true&quot; parse=&quot;false&quot;/&gt; <br>          &lt;/domains&gt;<br>          <br>          &lt;settings&gt;<br>            &lt;param name=&quot;debug&quot; value=&quot;1&quot;/&gt;<br>            &lt;param name=&quot;sip-trace&quot; value=&quot;no&quot;/&gt;<br>            &lt;param name=&quot;log-auth-failures&quot; value=&quot;false&quot;/&gt;<br>            &lt;param name=&quot;forward-unsolicited-mwi-notify&quot; value=&quot;false&quot;/&gt;<br>            &lt;param name=&quot;context&quot; value=&quot;asterisk&quot;/&gt;<br>            &lt;param name=&quot;rfc2833-pt&quot; value=&quot;101&quot;/&gt;<br>            &lt;param name=&quot;sip-port&quot; value=&quot;5050&quot;/&gt;<br>            &lt;param name=&quot;dialplan&quot; value=&quot;XML&quot;/&gt;<br>            &lt;param name=&quot;dtmf-type&quot; value=&quot;info&quot;/&gt;<br>            &lt;param name=&quot;inbound-codec-prefs&quot; value=&quot;$${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 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: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 class="h5">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:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><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>
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