<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>It turns out that I forgot to trigger SRST on the router itself so the gateways would register as H323. Once I did that, I get the output that I was used to seeing.<br><br>The TAC had a laugh as I fixed the problem myself while they were on the line. Well, that's they're "easy button" for the day.<br><br>I had thought about using a null route, but for some reason I couldn't get it working right. I had to deploy quickly the last time, and I'd like to keep the same format this time around. Something to think about in the future. I think the problem I had was that because it was not routeable, it didn't come back quickly with a call not progress signal, or something like that.<br><br>*shrug*<br><span><br><span name="x"></span>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it. <br> - LFJ (with apologies to Mr. Popeil)<br><span name="x"></span><br></span><br><hr id="zwchr"><b>From: </b>"VoIP Guy" <ciscovoiper1@gmail.com><br><b>To: </b>"Lelio Fulgenzi" <lelio@uoguelph.ca><br><b>Cc: </b>"Mathew Miller" <miller.mathew@gmail.com>, "voyp list" <cisco-voip@puck.nether.net><br><b>Sent: </b>Thursday, October 28, 2010 11:31:30 PM<br><b>Subject: </b>Re: [cisco-voip] help with debuging calls during SRST<br><br>I may be off base as some of the debug output is internal coding calls which I am not privy to but looking at it... it looks like it tries to match the dial-peer, it has an issue, and then tries again to match an incoming dial-peer... possibly because of the cor-list...<br>
<br>It would be helpful if you posted the cor list configuration, the relevant srst/cme config so we can determine how the lock & key method is setup in theory.<br><br>If all you want to do is to blackhole a call... just create a voip dial-peer, then get a subnet segment which is not routable on your network, like 192.168.x.x/172.16.x.x/10.x.x.x...<br>
<br>...create a route to null 0 for this specific route such as ip route 192.168.255.254 255.255.255.255 null0 description blackhole-voip<br><br>... then create a voip dial-peer as the above but without the cor-list and with the destination being ipv4:192.168.255.254, then add the command hunt stop and you should see the call get blackholed.<br>
<br>That is a very quick and dirty way of doing it... if you are after alternative suggestions, then I would suggest posting the requested info so we can take more of a look at it.<br><br>Also, it would be good for you to get a "baseline" state and hence remove the cor lists and see what the behaviour is without them.<br>
<br>Cheers,<br><br>C<br><br>Ps. Forgot to cc group so apologies Lelio for the duplicate email.<br><br><div class="gmail_quote">On Thu, Oct 28, 2010 at 8:25 PM, Lelio Fulgenzi <span dir="ltr"><<a href="mailto:lelio@uoguelph.ca" target="_blank">lelio@uoguelph.ca</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div><div style="font-family: Verdana; font-size: 10pt; color: rgb(0, 0, 0);">I need to block for specific users using COR lists. I don't think you can do that with after-hours.<br>
<br>I'd also like to stick with dial-peers for a few other reasons. <br><div class="im"><br><span><br><span></span>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it. <br> - LFJ (with apologies to Mr. Popeil)<br>
<span></span><br></span><br><hr></div><b>From: </b>"Mathew Miller" <<a href="mailto:miller.mathew@gmail.com" target="_blank">miller.mathew@gmail.com</a>><div class="im"><br><b>To: </b>"Lelio Fulgenzi" <<a href="mailto:lelio@uoguelph.ca" target="_blank">lelio@uoguelph.ca</a>><br>
</div><b>Cc: </b>"Leslie Meade" <<a href="mailto:lmeade@signal.ca" target="_blank">lmeade@signal.ca</a>>, "voyp list" <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
<b>Sent: </b>Thursday, October 28, 2010 1:54:37 PM<br><b>Subject: </b>Re: [cisco-voip] help with debuging calls during SRST<div><div></div><div class="h5"><br><br><div>Why not use afterhours block pattern.</div><div><br></div>
call-manager-fallback<div> after-hours block pattern 1 91900 7-24</div><div> after-hours day Sun 00:00 23:59</div><div><br></div><div><br><div><div>On Oct 28, 2010, at 12:10 PM, Lelio Fulgenzi wrote:</div><br><blockquote>
<div><div style="font-family: Verdana; font-size: 10pt; color: rgb(0, 0, 0);">actually, i don't want it to work, that's why i'm sending the BAD# prefix and no digits.<br><br>not the best way to prevent calls from going through, but the only one i know of right now.<span> </span><br>
<br>i just need to figure out which debugs to turn on to see what i'm sending. :(<br><span><br><span></span>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it.<span> </span><br> - LFJ (with apologies to Mr. Popeil)<br><span></span><br>
</span><br><hr><b>From:<span> </span></b>"Leslie Meade" <<a href="mailto:lmeade@signal.ca" style="color: blue; text-decoration: underline;" target="_blank">lmeade@signal.ca</a>><br><b>To:<span> </span></b>"Lelio Fulgenzi" <<a href="mailto:lelio@uoguelph.ca" style="color: blue; text-decoration: underline;" target="_blank">lelio@uoguelph.ca</a>>, "voyp list" <<a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline;" target="_blank">cisco-voip@puck.nether.net</a>><br>
<b>Sent:<span> </span></b>Thursday, October 28, 2010 12:47:12 PM<br><b>Subject:<span> </span></b>RE: [cisco-voip] help with debuging calls during SRST<br><br><div><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;">
<span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);">Add this line</span></div><p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;">
<span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);"> </span></p><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);">Forward-digits 3</span></div>
<p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);"> </span></p><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;">
<span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);">It will drop the 9 and send the rest..</span></div><p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;">
<span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 11pt; font-family: Calibri,sans-serif; color: rgb(31, 73, 125);"> </span></p><div>
<div style="border-width: 1pt medium medium; border-style: solid none none; padding: 3pt 0in 0in;"><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><b><span style="font-size: 10pt; font-family: Tahoma,sans-serif;">From:</span></b><span style="font-size: 10pt; font-family: Tahoma,sans-serif;"><span> </span><a href="mailto:cisco-voip-bounces@puck.nether.net" style="color: blue; text-decoration: underline;" target="_blank">cisco-voip-bounces@puck.nether.net</a><span> </span>[mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>]<span> </span><b>On Behalf Of<span> </span></b>Lelio Fulgenzi<br>
<b>Sent:</b><span> </span>Thursday, October 28, 2010 8:14 AM<br><b>To:</b><span> </span>voyp list<br><b>Subject:</b><span> </span>[cisco-voip] help with debuging calls during SRST</span></div></div></div><p class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;">
</p><div><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 10pt; font-family: Verdana,sans-serif; color: black;">I'm not having any luck debugging calls during SRST mode. What I'm looking to find out is:</span></div>
<ul style="margin-bottom: 0in;" type="disc"><li class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif; color: black;"><span style="font-size: 10pt; font-family: Verdana,sans-serif;">which dial-peer is being hit</span></li>
<li class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif; color: black;"><span style="font-size: 10pt; font-family: Verdana,sans-serif;">all the digits that are being sent out to the PRI (including translations and prefixes)</span></li>
<li class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif; color: black;"><span style="font-size: 10pt; font-family: Verdana,sans-serif;">which PRI is being used</span></li>
</ul><p class="MsoNormal" style="margin: 0in 0in 12pt; font-size: 12pt; font-family: 'Times New Roman',serif;"><span style="font-size: 10pt; font-family: Verdana,sans-serif; color: black;">I'm able to get which dial-peer is being hit and the translations, but I can't see the digits being sent to the PRI and which PRI is being used. I've got a crossover connected so both ports are up, but I don't think it's that, I think I'm just not using the right debug statements.<br>
<br>For example, when I hit the dial-peer below. I'd like to see the "BAD#" digits being sent to pri 0/0/0.</span></p><div style="margin-left: 30pt;"><div class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif; text-align: center;" align="center">
<span style="font-size: 10pt; font-family: Verdana,sans-serif; color: black;"><hr align="center" width="100%" size="2"></span></div><div style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif;">
<span style="font-size: 10pt; font-family: Verdana,sans-serif; color: black;">dial-peer voice 91811901 pots<br> corlist outgoing uogdev-block-services-css<br> huntstop<br> preference 2<br> destination-pattern 9[1-8]11<br>
clid network-number 5196741500<br> port 0/0/0:23<br> forward-digits 0<br> prefix BAD#</span></div><div class="MsoNormal" style="margin: 0in 0in 0.0001pt; font-size: 12pt; font-family: 'Times New Roman',serif; text-align: center;" align="center">
<span style="font-size: 10pt; font-family: Verdana,sans-serif; color: black;"><hr align="center" width="100%" size="2"></span></div></div><p class="MsoNormal" style="margin: 0in 0in 12pt; font-size: 12pt; font-family: 'Times New Roman',serif;">
<span style="font-size: 10pt; font-family: Verdana,sans-serif; color: black;"><br><br>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it.<span> </span><br> - LFJ (with apologies to Mr. Popeil)<br><br>
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