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<DIV dir=ltr><FONT color=#000000 size=2 face=Arial>Thanks! </FONT></DIV>
<DIV dir=ltr><FONT size=2 face=Arial></FONT> </DIV>
<DIV dir=ltr><FONT size=2 face=Arial>I'll do that --</FONT></DIV>
<DIV dir=ltr><FONT size=2 face=Arial></FONT> </DIV>
<DIV dir=ltr><FONT size=2 face=Arial>I think the PVDMs are OK, since they're getting regular use by the other PRI on the router that's in production.</FONT></DIV>
<DIV dir=ltr><FONT size=2 face=Arial></FONT> </DIV>
<DIV dir=ltr><FONT size=2 face=Arial>Tim</FONT></DIV></DIV>
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<FONT size=2 face=Tahoma><B>From:</B> Peter Slow [mailto:peter.slow@gmail.com]<BR><B>Sent:</B> Wed 4/4/2012 4:59 PM<BR><B>To:</B> Tim Reimers<BR><B>Cc:</B> cisco-voip@puck.nether.net<BR><B>Subject:</B> Re: [cisco-voip] Assistance with call routing question...<BR></FONT><BR></DIV>
<DIV>...Then you should use the isdn "service" command (i believe that's one of the keywords) under i THINK the serial/D channel interface to busy out channels and make sure you've tested every single one of them =)<BR>...Doing that still won't test all your DSPs depending on your hardware model, but Im assuming those are working unless you added new PVDMs with the PRI.<BR><BR>-Peter Slow<BR><BR><BR><BR>
<DIV class=gmail_quote>On Wed, Apr 4, 2012 at 3:51 PM, Tim Reimers <SPAN dir=ltr><<A href="mailto:treimers@ashevillenc.gov">treimers@ashevillenc.gov</A>></SPAN> wrote:<BR>
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<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt">Thanks! <U></U><U></U></SPAN></P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt"><U></U><U></U></SPAN> </P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt">That will do it!!<U></U><U></U></SPAN></P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt">For testing purposes, I certainly can just set up things so I have to dial “8” and then remove that using PreDot or a translation pattern, etc.<U></U><U></U></SPAN></P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt"><U></U><U></U></SPAN> </P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt">I’m only going to do this for a week or so for all my own calls (and those of anyone else I can guinea-pig into helping me test..)<U></U><U></U></SPAN></P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt">By then, I should have an idea of whether echo is a problem or not.<U></U><U></U></SPAN></P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt"><U></U><U></U></SPAN> </P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt">I’m testing carefully because this PRI is going to be used for a call-center, public-facing, and I really really don’t want to find out there are problems with it<U></U><U></U></SPAN></P>
<P class=MsoNormal><SPAN style="COLOR: #1f497d; FONT-SIZE: 11pt">by having customer service agents telling me that callers are angrily hanging up on the agent because they can’t hear them or the agent can’t hear the caller…<U></U><U></U></SPAN></P>
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<P class=MsoNormal><B><SPAN style="FONT-SIZE: 10pt">From:</SPAN></B><SPAN style="FONT-SIZE: 10pt"> Peter Slow [mailto:<A href="mailto:peter.slow@gmail.com" target=_blank>peter.slow@gmail.com</A>] <BR><B>Sent:</B> Wednesday, April 04, 2012 4:38 PM<BR><B>To:</B> Tim Reimers<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] Assistance with call routing question...<U></U><U></U></SPAN></P></DIV>
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<P class=MsoNormal>an h323 GW is all one entitiy from CUCMs perspective. If everything is in a single gateway, you'd have to use CUCM to prefic particular digits that the GW could then use to send to a particular interface.<BR><BR>How many PRIs are on this gateway, or are the other PRIs on _different_ gateways?<BR><BR>sounds like you're about to want to prefix somethign on the front of the called number, based on the callING number, which this document will show you how to do with translation profiles, directly in IOS:<U></U><U></U></P>
<H2>Mapping Outbound Calls to Unique FXS/FXO Ports on Analog Gateways<U></U><U></U></H2>
<P style="MARGIN-BOTTOM: 12pt" class=MsoNormal><A href="http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml" target=_blank>http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml</A><BR><BR><BR>-Peter Slow<BR><BR><BR><U></U><U></U></P>
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<P class=MsoNormal>On Wed, Apr 4, 2012 at 3:23 PM, Tim Reimers <<A href="mailto:treimers@ashevillenc.gov" target=_blank>treimers@ashevillenc.gov</A>> wrote:<U></U><U></U></P>
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<P class=MsoNormal>hi all-<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>I should be able to do this myself, but I am experiencing a BSOB (Blue screen of Brain) on this simple one—<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>I have a new PRI I want to test for echo and other issues—<U></U><U></U></P>
<P class=MsoNormal>For inbound calling, I already have a DID configured on the PRI and built on a couple of phones-<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>I’ve set up a couple of route patterns for specific numbers, and those calls do go out this PRI<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>For outbound calling, I’d like to simply set up a couple of phones in such a way that the only outside gateway they have access to is this PRI.<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>The gateway router is running this PRI in H.323 mode- <U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>I think I need to configure a new CSS in such a way that the only PSTN access for it would go through this one PRI?<U></U><U></U></P>
<P class=MsoNormal>but I’m stuck on that – <U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>For just one number, I can make the system use that PRI by doing this- <U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>dial-peer voice <myhomephone> pots<U></U><U></U></P>
<P class=MsoNormal>description Local call test on new PRI<U></U><U></U></P>
<P class=MsoNormal>destination-pattern 9<myhomephone><U></U><U></U></P>
<P class=MsoNormal>progress_ind setup enable 3<U></U><U></U></P>
<P class=MsoNormal>progress_ind alert enable 8<U></U><U></U></P>
<P class=MsoNormal>fax rate disable<U></U><U></U></P>
<P class=MsoNormal>port 0/1/0:23<U></U><U></U></P>
<P class=MsoNormal>forward-digits 7<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>If I expand that dial-peer out to wildcards, it’s going to affect everyone on the voice network, not just my test phones…<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>But I’m not certain how to set things up so that –all- calls to and from a couple of test phones will always use that one H.323 PRI.<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>There’s another PRI on that same gateway that’s in production, so I can’t just switch the config on the entire gateway at large—<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>When I try building a route list, all I can do is select that router and “all ports” – unlike MGCP, I don’t have the granular selection of just ports belonging to one Serial interface…<U></U><U></U></P>
<P class=MsoNormal><U></U><U></U> </P>
<P class=MsoNormal>Clues, anyone? <U></U><U></U></P></DIV></DIV>
<P style="MARGIN-BOTTOM: 12pt" class=MsoNormal><BR>_______________________________________________<BR>cisco-voip mailing list<BR><A href="mailto:cisco-voip@puck.nether.net" target=_blank>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><U></U><U></U></P></DIV>
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