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try adding 'incoming called number .' on your VOIP dial-peer<br>
<br>
<a class="moz-txt-link-freetext" href="http://www.cisco.com/en/US/docs/ios/12_2/voice/configuration/guide/vvfpeers.html">http://www.cisco.com/en/US/docs/ios/12_2/voice/configuration/guide/vvfpeers.html</a><br>
<br>
/wes<br>
<br>
On Thursday, August 27, 2009 4:47:57 PM , Sean Walberg
<a class="moz-txt-link-rfc2396E" href="mailto:sean@ertw.com"><sean@ertw.com></a> wrote:<br>
<blockquote
cite="mid:cc43a8f80908271347l4c7ad7ceu56a6a99e292de65@mail.gmail.com"
type="cite">Hi Wes,
<div><br>
</div>
<div>Thanks. "show call active voice" shows the IP leg is using
dial-peer 0 (PeerId=0) </div>
<div><br>
</div>
<div>I'm unclear how I should proceed though. Do I make a dial peer
that matches the calling number (eg, the CTI port) and turn DTMF relay
on that?</div>
<div><br>
</div>
<div>Thanks,</div>
<div><br>
</div>
<div>Sean</div>
<div><br>
<div class="gmail_quote">On Thu, Aug 27, 2009 at 3:16 PM, Wes Sisk <span
dir="ltr"><<a moz-do-not-send="true" href="mailto:wsisk@cisco.com">wsisk@cisco.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
most likely you are matching dial-peer 0 which does not use
dtmf-relay. setup a call and look at the call on the gateway to
confirm dial-peers in use.<br>
<br>
/wes
<div>
<div class="h5"><br>
<br>
On Thursday, August 27, 2009 3:14:50 PM , Sean Walberg
<a moz-do-not-send="true" href="mailto:sean@ertw.com"
target="_blank"><sean@ertw.com></a> wrote:<br>
</div>
</div>
<blockquote type="cite">
<div>
<div class="h5">I've got an IPCCX 5.0 SR2 system connected to my
Call
Manager 6.1 cluster. The subscriber, PSTN router, and IPCCX server are
all on the same switch.
<div><br>
</div>
<div>I've got a script that records a message from a caller,
makes an
outbound call to a defined phone to play back the message, and then
presents a menu to confirm the recipient got the message.</div>
<div><br>
</div>
<div>If the recipient of the message is an IP phone using the
internal extension, it works. IPCCX calls the phone, plays back the
message, and then the phone can press 2 to confirm.</div>
<div><br>
</div>
<div>If the recipient of the message goes out the PSTN (cell, or
even
calling the same IP phone by looping in and out the PSTN), the digits
aren't processed by IPCCX. You hear the message, but the digit
collection from the menu times out. "are you still there?" etc.</div>
<div><br>
</div>
<div>If I make a call in to IPCCX, digit collection works fine
even
from the same phone that doesn't work if IPCCX calls it.</div>
<div><br>
</div>
<div>The gateway is H.323 and dtmf relaying is on. Packet traces
show
that the digits seem to be going in band and we can hear the digits
being played over a G.711 stream.</div>
<div><br>
</div>
<div>Nothing in the bug toolkit, and Google always leads me to
Unity
call handler problems and dtmf relay being disabled)</div>
<div><br>
</div>
<div>I've poked around the CTI configuration and don't see where
I
could set up any DTMF relaying on the IPCCX side. There is a media
termination point being used though.</div>
<div><br>
</div>
<div>Any ideas where to look next?</div>
<div><br>
</div>
<div>Thanks,</div>
<div><br>
</div>
<div>Sean</div>
<div><br>
</div>
<div><br>
</div>
<div>-- <br>
Sean Walberg <<a moz-do-not-send="true" href="mailto:sean@ertw.com"
target="_blank">sean@ertw.com</a>>
<a moz-do-not-send="true" href="http://ertw.com/" target="_blank">http://ertw.com/</a><br>
</div>
</div>
</div>
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</blockquote>
<br>
</div>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
Sean Walberg <<a moz-do-not-send="true" href="mailto:sean@ertw.com">sean@ertw.com</a>>
<a moz-do-not-send="true" href="http://ertw.com/">http://ertw.com/</a><br>
</div>
</blockquote>
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