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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Eric and Jim –<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Your (Eric) situation sounds like MTP resources may be running
out. When CUCM is in the loop – my guess you are using
software MTPs to handle DTMF, etc negotiations. By looking in your
CUCM trace files will indicate if this is the case. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Two key reasons SIP has problems:<o:p></o:p></span></p>
<p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span
style='mso-list:Ignore'>1.<span style='font:7.0pt "Times New Roman"'>
</span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>EO versus DO – early offer versus delayed
offer EO requires MTP on CUCM at this time and EO is
many times what telco’s and some vendors require.<o:p></o:p></span></p>
<p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span
style='mso-list:Ignore'>2.<span style='font:7.0pt "Times New Roman"'>
</span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>DTMF - the “norm” is leaning to RFC2833 –
though some companies still use other methods. However, if your
applications do not support RFC2833 you will require MTPs. The phone loads,
Unity(as I recall), etc do support it. In addition, if you are
running MGCP gateways and have SIP phones, there are commands to allow MGCP to
speak RFC2833.<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'>As a side note:<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>If you are hitting any issues with transfers, conferencing, etc…
simply add router-based software MTPs on the CUBE or ingress gateway and it
will resolve most if not all of these issues.<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'>Nick covers some items below. A couple of others I
have seen cause issues:<o:p></o:p></span></p>
<p class=MsoListParagraph style='text-indent:-.25in;mso-list:l1 level1 lfo2'><![if !supportLists]><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span
style='mso-list:Ignore'>1.<span style='font:7.0pt "Times New Roman"'>
</span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Make sure that the CUCM Group you have assigned via the device
pool on the SIP trunks matches the Subscribers that you are pointing to with
your dial peers. That is if your dial peers point to Subs 1, 2 and
4. Set up your group for 1,2 and 4. <o:p></o:p></span></p>
<p class=MsoListParagraph style='text-indent:-.25in;mso-list:l1 level1 lfo2'><![if !supportLists]><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span
style='mso-list:Ignore'>2.<span style='font:7.0pt "Times New Roman"'>
</span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Transcoding and DTMF – validate that if you are using MTPs
that they can handle what you need them to. Different MTP sources
can handle only specific items – see the SRND, etc for more detail.<o:p></o:p></span></p>
<p class=MsoListParagraph style='text-indent:-.25in;mso-list:l1 level1 lfo2'><![if !supportLists]><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><span
style='mso-list:Ignore'>3.<span style='font:7.0pt "Times New Roman"'>
</span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Nick mentions it below but do make sure your pTime is the same
(packetization time). Early on when integrating CUCM 6.X and Microsoft UM
this was an issue depending on how MTPs were used.<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'>Lastly, if you want to reach out to me, I am more than willing
to jump on a call to chat.<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'>Cheers,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Bob <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Solutions Architect<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Customer Interactive Solutions<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Dimension Data<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'><a href="www.dimensiondata.com">www.dimensiondata.com</a><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 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>Nick Matthews<br>
<b>Sent:</b> Monday, September 27, 2010 6:36 PM<br>
<b>To:</b> Eric Brander<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk question<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>95% of the time these are dial
peer misconfigurations. Run 'debug voip dialpeer' in your two situations
with no other calls on the router, and see if there is a difference. If
you hit dial peer 0, you need to read about incoming dial peers and fix the problem.
Another 3% are just a interop issue and you need to reconfigure the DTMF method
from end-to-end, or somewhere in the middle.<br>
<br>
In very rare situations timing issues prevent DTMF from happening.
Changing H.323 to slow-start to fast-start or vice versa will sometimes fix
that, or using an MTP on CUCM.<br>
<br>
-nick<o:p></o:p></p>
<div>
<p class=MsoNormal>On Mon, Sep 27, 2010 at 6:40 PM, Eric Brander <<a
href="mailto:mailinglists@rednarb.com">mailinglists@rednarb.com</a>> wrote:<o:p></o:p></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'>On Mon, Sep 27, 2010 at 2:53
PM, Jim McBurnett <<a href="mailto:jim@tgasolutions.com">jim@tgasolutions.com</a>>
wrote:<br>
> So does anyone have a good idea as to why some calls on a SIP trunk will<br>
> work fine with DTMF and a few fail?<br>
><br>
><br>
><br>
> IE—<br>
><br>
> Call to X – AA options don’t work..<br>
><br>
> Call to Y—work fine..<br>
><o:p></o:p></p>
</div>
<p class=MsoNormal>I'm fighting with sometjhing similar right now in a
configuration<br>
between Cisco and NobleSystems (dialer platform based on Asterisk). If<br>
the call placed from my Cisco desk phone communicates directly to a<br>
call manager which then in turn passes the data across the SIP trunk<br>
to Noble then everything works fine. Sometimes however the call will<br>
be handled initially by the call manager but then the phone<br>
communicates directly to the Noble SIP server instead of using the<br>
call manager as an intermediary - when this happens it appears not all<br>
commands are accepted, including DTMF. It seems load related - the<br>
more SIP calls in-place, the more likely for this direct connection<br>
problem.<br>
<br>
I can't for the life of me find a solution... but not sure if its<br>
related to your issue or not but sure sounds similar.<br>
<br>
_______________________________________________<br>
cisco-voip mailing list<br>
<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
<a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
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