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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Again thanks for the additional information. Because I don’t want
to solely limit myself to MGCP knowledge, does the same go for H.323 other then
RDNIS has to be enabled at the gateway rather than in CCM.<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"'> Justin Steinberg
[mailto:jsteinberg@gmail.com] <br>
<b>Sent:</b> Tuesday, May 08, 2007 10:08 AM<br>
<b>To:</b> Linsemier, Matthew<br>
<b>Subject:</b> Re: [cisco-voip] Voicemail in SRST Mode<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>no, DID will not cause any
problem. The extension of the SRST phone is used as the redirecting
number. Hopefully, your extension is the last 3,4, or 5 digits of your
DID.<br>
<br>
Easiest way to understand how this works is to do a 'debug isdn q931' while in
SRST mode and place a call from the outside to an IP phone and watch the
outbound fwd leg to unity. <o:p></o:p></p>
<div>
<p class=MsoNormal><span class=gmailquote>On 5/8/07, <b>Linsemier, Matthew</b>
<<a href="mailto:MLinsemier@apcapital.com">MLinsemier@apcapital.com</a>>
wrote:</span><o:p></o:p></p>
<div>
<div>
<p><span style='font-size:11.0pt;color:#1F497D'>Justin,</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Thanks for the response. I
also should have mentioned that we have DID at all of our sites. From
what you are saying, it sounds like this could cause problems because there
would be no redirecting number?</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Matt</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;
border-color:-moz-use-text-color -moz-use-text-color'>
<p><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:
10.0pt'> Justin Steinberg [mailto:<a href="mailto:jsteinberg@gmail.com"
target="_blank">jsteinberg@gmail.com</a>] <br>
<b>Sent:</b> Tuesday, May 08, 2007 9:55 AM<br>
<b>To:</b> Linsemier, Matthew<br>
<b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] Voicemail in SRST Mode</span><o:p></o:p></p>
</div>
<div>
<p> <o:p></o:p></p>
<p style='margin-bottom:12.0pt'>If you have a PRI at both the SRST & Unity
site the easiest integration is using RDNIS. However, not all Telco
providers support end to end RDNIS. Check to see if yours does by
debugging q931 on both your SRST site and your Unity site. <br>
<br>
You will see a debug similiar to below when the SRST router forwards the call
to <br>
<br>
Calling Party Number i = 0x0081, '5555553435'<br>
Plan:Unknown, Type:Unknown<br>
Called Party Number i = 0x80, '15555551212' <br>
Plan:Unknown, Type:Unknown <br>
Redirecting Number i = 0x000082, '1001'<br>
Plan:Unknown, Type:Unknown <br>
<br>
Where 5555553435 is a PSTN caller who is calling SRST extension 1001 and the
call is forwarded to Unity pilot 5555551212.<br>
<br>
You should then perform the same debug on the Unity PRI gateway to see if the
RDNIS comes through, if so then you just need to make sure the RDNIS number
matches the Unity extension and via ccmadmin enable inbound RDNIS on the MGCP
T1 PRI at the Unity site <br>
<br>
If your Telco doesn't support RDNIS then you will have to use vm-integration
which is typically used for FXO lines but can also be used for pri.<br>
<br>
Justin<o:p></o:p></p>
<div>
<p>On 5/8/07, <b>Linsemier, Matthew</b> <<a
href="mailto:MLinsemier@apcapital.com" target="_blank">MLinsemier@apcapital.com</a>>
wrote:<o:p></o:p></p>
<div>
<div>
<p><span style='font-size:11.0pt;color:#1F497D'>Currently we are running
CallManager 4.1(3)sr3 along with Unity 4.2(1). We currently are using
MGCP to control all of our gateways from a centralized cluster and utilize SRST
at remote sites for failover. I was wondering if anyone has been
successful, while in SRST mode, of redirecting an inbound call after a set
amount of rings to their voicemail server, and have the subscribers message
play rather than the main Unity greeting. We are using ISDN PRI at our
remote sites. I know that there were some commands that you could configure to
send digits to the Unity server, but from what I understand those were only
supported in a analog configuration.</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Below is a excerpt of my
call-manager fallback config:</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p style='margin-left:.5in'><span style='font-size:11.0pt;color:#1F497D'>call-manager-fallback</span><o:p></o:p></p>
<p style='margin-left:.5in'><span style='font-size:11.0pt;color:#1F497D'> voicemail
91517xxxxxxx</span><o:p></o:p></p>
<p style='margin-left:.5in'><span style='font-size:11.0pt;color:#1F497D'> call-forward
busy 91517xxxxxxx</span><o:p></o:p></p>
<p style='margin-left:.5in'><span style='font-size:11.0pt;color:#1F497D'> call-forward
noan 9517xxxxxxx timeout 12</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Any help would be greatly
appreciated.</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>Matt</span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
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