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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>This is a collection of feedback from various sources and a 5
hour TAC case this week and should be peer reviewed and researched further
(never take anything I type as the given final word). More importantly this is
for Google next time I go down this path!<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'>1. This
was for a Verizon SIP Trunk to 12.4(24)T CUBE (SIP-SIP) to CallManager
7.0(2) to UCCX 7.0(1)SR1 to Cisco Agent Desktop. Callers couldn’t
input DTMF in the UCCX aef scripts, Agents couldn’t Transfer blind were
symptoms.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>2. In
IOS running-cnfig “dtmf-relay rtp-nte” is actually RFC 2833
which encodes the DTMF key presses and other telephony events as part of the
RTP (audio) stream of the SIP call. RFC2833 is what Verizon is sending/using.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>3. Dtmf-relay
rtp-nte digit-drop requires a MTP resource when going to UCCX 7x or Unity 7x,
it can be a software MTP resource on a router. Without MTP resources the
menu input options in a aef script won’t capture caller input. Item 3 isn't
100% accurate in every scenario as there are plenty of devices that support
RFC2833 for DTMF (MGCP gateways if properly configured, IP phones, etc) and
thus no MTP would be required for DTMF negotiation between those
endpoints. DTMF negotiation is much like RTP codec negotiation in
that if both parties do not support the same DTMF method than an MTP (or
transcoder) is required to allow the call to proceed.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>4. A
3845 router should be setup for no more than 500 software MTP resources
(remember one call has two legs, so it’s two resources per call) and put
it its own MRG above CallManager software MTP resources in your MRGL. The 500
number is CPU dependant, watch your show proc cpu history. Issue a show sccp
all to verify you aren’t using up all the resources.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>5. Don’t
put a checkmark under Device > Trunk for MTP Resource Required, as
CallManager 6x/7x will dynamically allocate MTP resources without it
checked. Otherwise if checked you are forcing every call to use MTP
whether it needs it or not and you likely don’t have enough resources for
that (or you are wasting resources needlessly).<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>6. Use
codec-passthrough under dspfarm to reduce router CPU usage;<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>dspfarm
profile 11 mtp<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> codec
pass-through<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> maximum
sessions software 500<o:p></o:p></span></p>
<p class=MsoNormal style='text-indent:.5in'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'> associate application
SCCP<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>7. Use
RTMT to check for Media Resources Exhausted. Use show sccp all to see how
many MTP resources are available on 3845. Each call needing a dynamic MTP
would use two resources, one for each leg. So a 3845 with 500 software
MTP resources reserved would support 250 calls if those calls were determined
by CallManager dynamically to need a MTP resource.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>8. Verizon
Business SIP dial-peers being used for Fax Machines have to use G711 or
they will fail with G729 (which is Verizon default codec). Best bet is to
use local PRI/FXO resources for faxing or a T38 service or eFax.com, etc. Lots
of complaints on Cisco-voip mailing list and Asterisk Users mailing list about
faxing over Verizon Business SIP trunks. This customer spent 3 days trying to
get faxing to work over Verizon SIP trunk before they setup a FXO interface and
routed faxes out that.<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'>I need to research this further as I understood the recommend
method is SIP to CUBE and then CUBE to CallManager as H323, however I inherited
this and not sure anyone did a design recommendation to customer or if customer
pushed this design without experience or research, etc.<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'><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>Erin Mikelson<br>
<b>Sent:</b> Friday, May 29, 2009 5:38 PM<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] DTMF relay with Verizon SIP Trunk?<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;
font-family:"Verdana","sans-serif"'>A collegue of mine spent 32 hours on the
phone with TAC and Verzion due to a similar issue.<br>
<br>
I don't remember the specifics of what he was running into, but in short, this
is what he required:<br>
<br>
You need 12.4.22(T2)<br>
You need H.323 faststart which requires MTP<br>
You need to configure MTP resources in the router as software<br>
You need to make sure your MRGs are set to use the above MTP and Not any
other hardware transcoders or software MTPs.<br>
<br>
Good Luck?<br>
> <br>
> <br>
> ----------------------------------------------------------------------<br>
> <br>
> Message: 1<br>
> Date: Thu, 28 May 2009 12:02:13 -0400<br>
> From: "Jason Aarons (US)" <jason.aarons@us.didata.com><br>
> To: <cisco-voip@puck.nether.net><br>
> Subject: [cisco-voip] DTMF relay with Verizon SIP Trunk?<br>
> Message-ID:<br>
> <C1FE15183DA37645BC0633BC604E44F00EF56C88@USNAEXCH.na.didata.local><br>
> Content-Type: text/plain; charset="us-ascii"<br>
> <br>
> I assume this is the correct DTMF relay for a Verizon SIP trunk.<br>
> <br>
> <br>
> <br>
> Using CallManager 7.0(2), no MTP resources, SIP Trunk to CUBE 12.4(24)T<br>
> with SIP to Verizon.<br>
> <br>
> <br>
> <br>
> At first we had below, then I realized DTMF was working;<br>
> <br>
> dial-peer voice 797 voip<br>
> <br>
> no dtmf-relay h245-alphanumeric h245-signal cisco-rtp<br>
> <br>
> dtmf-relay rtp-nte digit-drop<br>
> <br>
> <br>
> <br>
> <br>
> <br>
> Does faxing/VG224 have any dtmf-relay requirements? I understand they<br>
> were using rtp-nte digit-drop but changed it for some unknown reason to<br>
> h245-alphanumic which I understand is only for H323.<br>
> <br>
> <br>
> <br>
> <br>
> <o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Hotmail®
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