<div>Hi Bob</div>
<div> </div>
<div>Ok, I have IOS transcoder and IOS MTP resources setup, where should I apply these, to the MGCP gateway. </div>
<div> </div>
<div>Regards</div>
<div> </div>
<div>Nick<br><br></div>
<div class="gmail_quote">On 15 February 2011 21:51, Bob Zanett (US) <span dir="ltr"><<a href="mailto:bob.zanett@us.didata.com">bob.zanett@us.didata.com</a>></span> wrote:<br>
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<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">Going between SIP and MGCP – you will most likely require an MTP resource. A transcoder will perform that but you may not require a full transcode resource. </span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">DTMF: Usually DTMF has to be converted from RFC2833 (typically used) to MGCP. There is a command you can place on the MGCP gateway to allow it to understand RFC 2833 DTMF – search on Cisco for MGCP and RFC2833. However, you would need to validate that all potential endpoints within your Cisco environment can also handle it – otherwise MTP again.</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">Codec: I would assume your codec is remaining the same either G711 or G729. If codec changes, then yes you would require a transcode resource.</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
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<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">Bob </span></p></div>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
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<p class="MsoNormal"><b><span style="FONT-SIZE: 10pt">From:</span></b><span style="FONT-SIZE: 10pt"> <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>ROZA, Ariel<br>
<b>Sent:</b> Tuesday, February 15, 2011 3:19 PM<br><b>To:</b> Nick; <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> Re: [cisco-voip] Inbound SIP Calls routing over QSIG link</span></p>
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<div></div>
<div class="h5">
<p class="MsoNormal"> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt" lang="ES-AR">Nick,</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt" lang="ES-AR"> </span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">A Resources Unavailable message sounds like you need to enable the use of transcoding for the call.</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><b><span style="FONT-SIZE: 10pt">From:</span></b><span style="FONT-SIZE: 10pt"> <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>Nick<br>
<b>Sent:</b> martes, 15 de febrero de 2011 01:24 p.m.<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] Inbound SIP Calls routing over QSIG link</span></p>
<p class="MsoNormal"><span lang="ES-AR"> </span></p>
<div>
<p class="MsoNormal"><span lang="ES-AR">Hi All</span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR"> </span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR">I'm trying to route inbound calls from SIP trunks through CUCM to another PBX connected with QSIG, I have CUCM 8.0.3, with QSIG links to a IPC dealerboard system and also a Nortel CS1000 PBX, these are set up as MGCP in CUCM.</span></p>
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<p class="MsoNormal"><span lang="ES-AR"> </span></p></div>
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<p class="MsoNormal"><span lang="ES-AR">I also have SIP trunks for PSTN providing both inbound and outbound calling. The issue I have is when I route inbound calls from the SIP trunks to CUCM and across the QSIG links to the PBX's, the call will not complete and I get Resources Unavailable, Unspecified on the debug on the gateway, I have also taken a SIP trace and get a 503 Service Unavailable from CUCM to the SIP trunks for the same call.</span></p>
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<div>
<p class="MsoNormal"><span lang="ES-AR"> </span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR">Has anyone successfully done this before and can I route SIP to Qsig or would I need a CUBE even though its going through CUCM.</span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR"> </span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR">I have attached the debug and SIP trace.</span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR"> </span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR">Any help would be greatly appreciated as this is rather urgent now.</span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR"> </span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR">Regards</span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR"> </span></p></div>
<div>
<p class="MsoNormal"><span lang="ES-AR">Nick </span></p></div></div></div></div></div>
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