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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Bob <o:p></o:p></span></p>
</div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></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;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>ROZA, Ariel<br>
<b>Sent:</b> Tuesday, February 15, 2011 3:19 PM<br>
<b>To:</b> Nick; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Inbound SIP Calls routing over QSIG link<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span lang=ES-AR style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Nick,<o:p></o:p></span></p>
<p class=MsoNormal><span lang=ES-AR style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>A Resources Unavailable message sounds like you need to enable
the use of transcoding for the call.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></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-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Nick<br>
<b>Sent:</b> martes, 15 de febrero de 2011 01:24 p.m.<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> [cisco-voip] Inbound SIP Calls routing over QSIG link<o:p></o:p></span></p>
<p class=MsoNormal><span lang=ES-AR><o:p> </o:p></span></p>
<div>
<p class=MsoNormal><span lang=ES-AR>Hi All<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR> <o:p></o:p></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.<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR> <o:p></o:p></span></p>
</div>
<div>
<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.<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR> <o:p></o:p></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.<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR> <o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR>I have attached the debug and SIP trace.<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR> <o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR>Any help would be greatly appreciated as
this is rather urgent now.<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR> <o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR>Regards<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR> <o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span lang=ES-AR>Nick <o:p></o:p></span></p>
</div>
</div>
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