<div dir="ltr"><div><div><div>hello guys,<br><br></div>thank you for your answers. i have a voice class codec and assign it to both dial-peers. cisco receives calls with g711alaw codec and must transcode it to g711ulaw because the freepbx which cisco is connected to in the other side, supports just g711ulaw codec. my question is: can cisco 2800 transcode these codecs? if yes, is it enabled by default or should i enable it by configuring "dspfarm profile"?<br><br></div>thanks you,<br></div>SAM<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Apr 2, 2015 at 3:06 PM, <span dir="ltr"><<a href="mailto:Zoltan.Kelemen@emerson.com" target="_blank">Zoltan.Kelemen@emerson.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div link="blue" vlink="purple" lang="EN-US">
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Hi Sam,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Do you need to transcode on the Cisco? (i.e. aren’t both endpoints supporting the same set of codecs?)
<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">You need to ensure, that if you use a codec list on one call-leg (dial-peer) you support the same on the other call-leg (or outbound dial-peer) as well.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Trying transparent codec on dial-peers may also work.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">For transcoding you’ll need DSCP resources and some configuration:<br>
<a href="http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/transcoding.html" target="_blank">http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/transcoding.html</a><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Cheers,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Arial","sans-serif";color:#0f245f">Zoltan Kelemen</span></b><b><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#0f245f"><br>
</span></b><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#666666">Emerson</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> cisco-voip [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>s m<br>
<b>Sent:</b> 02 April 2015 12:08<br>
<b>To:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><span class=""><br>
<b>Subject:</b> [cisco-voip] sip trunk to asterisk<u></u><u></u></span></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">hello everybody,<u></u><u></u></p>
</div><div><div class="h5">
<p class="MsoNormal" style="margin-bottom:12.0pt">i want to configure a sip trunk between a cisco router and my system which has asterisk. this is my scenario:<u></u><u></u></p>
</div></div></div>
<p class="MsoNormal" style="margin-bottom:12.0pt">Freepbx-----my system-----cisco-router----Freepbx<u></u><u></u></p>
</div><div><div class="h5">
<p class="MsoNormal">my system acts like a router. in cisco, if i set just one codec in dial-peers, every thing is ok and i can make a call. but if i set different codecs in a voice class codec and assign it to dial-peers, i can make call but call is terminated.<u></u><u></u></p>
</div></div></div><div><div class="h5">
<p class="MsoNormal" style="margin-bottom:12.0pt">i trace all debug messages in cisco and think it is happened when receiving call has a codec which differs with first codec priority in cisco router because my cisco router can not transcode these codecs to
each other. am i right or misunderstand?? if it is true, how can i enable transcoding on my cisco router? i have a 2800 router.<u></u><u></u></p>
</div></div></div><div><div class="h5">
<p class="MsoNormal">any comments ot hints are really appreciated.<u></u><u></u></p>
</div></div></div>
<p class="MsoNormal">SAM<u></u><u></u></p>
</div>
</div>
</div>
</blockquote></div><br></div>