Call comes in local VGW (MGCP) one of 5 PRI's, all in the G711 Region.<div><br></div><div><div>MTP's are the Pub and Sub. Both are registered and working.</div><div><br></div><div>Scott</div><div><br><div class="gmail_quote">
On Wed, Sep 22, 2010 at 11:14 AM, Jason Aarons (US) <span dir="ltr"><<a href="mailto:jason.aarons@us.didata.com">jason.aarons@us.didata.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">To where are they transferring and what are codecs/regions. You have to be able to reproduce the problem or look thru logs like a needle in a haystack. </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Transfer invokes a hold, then Transfers, check a ccm trace of a failed call. Either lack of transcoding resources possibly, lack of MTP resources, etc. Does RTMT shows any resource (MTP) issues?</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Ideally you don’t want calls using a MTP for every call.</span></p><p class="MsoNormal">
<span style="font-size:11.0pt;color:#1F497D"> </span></p><div><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>James Buchanan<br>
<b>Sent:</b> Wednesday, September 22, 2010 2:10 PM<br><b>To:</b> Scott Voll; <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> Re: [cisco-voip] How does Transfer work?</span></p>
</div></div><div><div></div><div class="h5"><p class="MsoNormal"> </p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Are you using a SIP trunk?</span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p class="MsoNormal"><b><span style="font-size:7.0pt;color:#7F7F7F">James Buchanan </span></b><span style="font-size:7.0pt;color:#7E7E7E">|</span><b><span style="font-size:7.0pt;color:#7F7F7F"> Senior Network Engineer </span></b><span style="font-size:7.0pt;color:#7E7E7E">|</span><span style="font-size:7.0pt;color:#7F7F7F"> <b>South Region </b></span><span style="font-size:7.0pt;color:#7E7E7E">|</span><b><span style="font-size:7.0pt;color:#7F7F7F"> Presidio Networked Solutions <br>
12 Cadillac Dr, Suite 130, Brentwood, TN 37027 </span></b><span style="font-size:7.0pt;color:#7E7E7E">|</span><b><span style="font-size:7.0pt;color:#7F7F7F"> </span></b><b><span style="font-size:7.0pt;color:#7E7E7E"><a href="mailto:jbuchanan@ctiusa.com" target="_blank">jbuchanan@presidio.com</a><br>
</span></b><b><span style="font-size:7.0pt;color:#7F7F7F">D: 615-866-5729</span></b><span style="font-size:7.0pt;color:#7F7F7F"> </span><span style="font-size:7.0pt;color:#7E7E7E">| <b>F:</b> </span><b><span style="font-size:7.0pt;color:#7F7F7F">615-866-5781</span></b><span style="font-size:7.0pt;color:#7F7F7F"> | </span><b><span style="font-size:7.0pt;color:#7E7E7E"><a href="http://www.presidio.com" target="_blank">www.presidio.com</a></span></b><span style="font-size:7.0pt;color:#1F497D"> <</span><b><span style="font-size:7.0pt;color:#7E7E7E"><a href="http://www.presidio.com" target="_blank">http://www.presidio.com</a></span></b><span style="font-size:7.0pt;color:#1F497D">> </span></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><b><span style="font-size:7.0pt;color:#7F7F7F">CCIE #25863, Voice</span></b><span style="font-size:11.0pt;color:#1F497D"></span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Scott Voll<br>
<b>Sent:</b> Wednesday, September 22, 2010 12:55 PM<br><b>To:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> [cisco-voip] How does Transfer work?</span></p>
<p class="MsoNormal"> </p><p class="MsoNormal">Sounds really stupid, I know.</p><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">But how does a transfer work?</p></div><div><p class="MsoNormal"> </p></div><div>
<p class="MsoNormal">With a Conference, it uses the Conference resource.</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">We are having intermittent calls drop during a transfer and I'm trying to find an all in compassing reason.</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">originally I thought it was a soft phone issue. but it looks to be happening in house too.</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">
So even on a completely G711 call I'm getting calls drop. Unfortunately I don't have Caller ID on the incoming calls so it's making it even harder to troubleshoot.</p></div><div><p class="MsoNormal"> </p></div>
<div><p class="MsoNormal">I can NOT reproduce on demand but it's happening frequently enough to try and get to the bottom of it.</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">TIA</p></div><div>
<p class="MsoNormal"> </p></div><div><p class="MsoNormal">Scott </p></div></div></div></div></div>
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