<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>Just as a followup, the 'default-destination' did not work with the PRI. According to online docs, I think this is for FXO only. So I removed the dial-peer and edited the voice port as I mentioned below to create a PLAR.<br><br>Thanks for everyone's help.<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 (ANNU)<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>"Lelio Fulgenzi" <lelio@uoguelph.ca><br><b>To: </b>"Nick Matthews" <matthnick@gmail.com><br><b>Cc: </b>"Nate VanMaren" <VanMarenNP@ldschurch.org>, "cisco-voip@puck.nether.net (cisco-voip@puck.nether.net)" <cisco-voip@puck.nether.net><br><b>Sent: </b>Wednesday, June 6, 2012 4:57:04 PM<br><b>Subject: </b>Re: [cisco-voip] dial peer/config to send all calls to one number<br><br><style>p { margin: 0; }</style><div style="font-family: Verdana; font-size: 10pt; color: #000000">Thanks Nick. I looked up an existing router, and it was very similar to what you have presented.<br><br>Someone offline also suggested using the 'default-destination' command under the 'call-manager-fallback' section, so I'm gonna try that first and see if it works.<br><br>First have to set up a T1 cross over connection from my 6608s to the PRIs on the router. I want to do this so I don't have to do maintenance on our trunks to test inbound calls.<br><br>Lelio<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 (ANNU)<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 id="zwchr"><b>From: </b>"Nick Matthews" <matthnick@gmail.com><br><b>To: </b>"Lelio Fulgenzi" <lelio@uoguelph.ca><br><b>Cc: </b>"Nate VanMaren" <VanMarenNP@ldschurch.org>, "cisco-voip@puck.nether.net (cisco-voip@puck.nether.net)" <cisco-voip@puck.nether.net><br><b>Sent: </b>Wednesday, June 6, 2012 4:53:29 PM<br><b>Subject: </b>Re: [cisco-voip] dial peer/config to send all calls to one number<br><br>(from memory, may not be 100%)<br><br>dial-peer voice 2 pots<br>port 0/0/0:23<br>direct-inward-dial<br>translation-profile incoming SET12345<br><br>dial-peer voice 1 voip<br>destination-pattern 12345<br>session target ipv4:X.X.X.X<br>etc<br><br>voice translation-profile SET12345<br>translate called 1<br><br>voice translation-rule 1<br>rule 1 /.*/ /12345/<br><br>To change the number all you do is go back to voice translation-rule 1<br>and change rule 1 to something else.<br><br>-nick<br><br>On Wed, Jun 6, 2012 at 10:47 AM, Lelio Fulgenzi <lelio@uoguelph.ca> wrote:<br>> it's more for a matter of getting it done quickly for testing ACLs, not for<br>> getting things working in a production environment. a one line config would<br>> work better for me. ;)<br>><br>> but i see your point.<br>><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 (ANNU)<br>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>> Cooking with unix is easy. You just sed it and forget it.<br>> - LFJ (with apologies to Mr. Popeil)<br>><br>><br>> ________________________________<br>> From: "Nate VanMaren" <VanMarenNP@ldschurch.org><br>> To: "Lelio Fulgenzi" <lelio@uoguelph.ca>, "cisco-voip@puck.nether.net<br>> (cisco-voip@puck.nether.net)" <cisco-voip@puck.nether.net><br>> Sent: Wednesday, June 6, 2012 10:41:48 AM<br>> Subject: RE: [cisco-voip] dial peer/config to send all calls to one number<br>><br>><br>> “i'd like to avoid having to setup translations and the like.” Why?<br>><br>><br>><br>> On every gateway I setup these basic things, and adjust for the country.<br>> Some sites there is a direct mapping between DID and extension, some it’s<br>> random. I do that all on the gateway so it will work in SRST too. I have<br>> moved from H.323 to SIP for all new gateways, but there is some stuff in<br>> here that allows for both. (CM h.323 doesn’t send the +)<br>><br>><br>><br>> I gave up on trying to use MGCP on a SRST box for the trunks a long time<br>> ago. I do use MGCP for the FXS ports, because SCCP doesn’t do Protocol<br>> based T.38.<br>><br>><br>><br>><br>><br>> trunk group PSTN<br>><br>> translation-profile incoming FROMPSTN<br>><br>> translation-profile outgoing TOPSTN<br>><br>> voice translation-rule 10<br>><br>> rule 1 /\(^9994870120\)/ /+1\1/<br>><br>> rule 100 /\(^9995472[23]..\)/ /+1\1/<br>><br>> !<br>><br>> voice translation-rule 11<br>><br>> rule 1 /\(.*\)/ /+1\1/ type national national<br>><br>> rule 2 /\(.*\)/ /+\1/ type international international<br>><br>> !<br>><br>> voice translation-rule 15<br>><br>> rule 1 /^1/ // type any national plan any isdn<br>><br>> rule 2 /^\+1/ // type any national plan any isdn<br>><br>> rule 4 /^\+/ // type any international plan any isdn<br>><br>> !<br>><br>> voice translation-rule 16<br>><br>> rule 1 /^011/ /011/ type any unknown plan any unknown<br>><br>> rule 2 /^9011/ /011/ type any unknown plan any unknown<br>><br>> rule 3 /^\+1/ /1/ type any national plan any isdn<br>><br>> !<br>><br>> !<br>><br>> voice translation-profile FROMPSTN<br>><br>> translate calling 11<br>><br>> translate called 10<br>><br>> !<br>><br>> voice translation-profile TOPSTN<br>><br>> translate calling 15<br>><br>> translate called 16<br>><br>><br>><br>> From: cisco-voip-bounces@puck.nether.net<br>> [mailto:cisco-voip-bounces@puck.nether.net] On Behalf Of Lelio Fulgenzi<br>> Sent: Wednesday, June 06, 2012 8:26 AM<br>> To: cisco-voip@puck.nether.net (cisco-voip@puck.nether.net)<br>> Subject: [cisco-voip] dial peer/config to send all calls to one number<br>><br>><br>><br>><br>> i'm trying to do some testing and I'd like something simple to send all<br>> calls that come in from a PRI during SRST (i.e. H323) to one extension.<br>><br>> Would I use a voice-port config as follows?<br>><br>> voice-port 0/0/1:23<br>> connection plar 12345<br>><br>> -or- would I use a dial-peer config as follows?<br>><br>> dial-peer voice 902 pots<br>> incoming called-number .<br>> direct-inward-dial<br>> port 0/0/1:23<br>><br>> if dial-peer, what other command would I use to send the call to extension<br>> 12345?<br>><br>> i'd like to avoid having to setup translations and the like.<br>><br>><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 (ANNU)<br>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>> Cooking with unix is easy. You just sed it and forget it.<br>> - LFJ (with apologies to Mr. Popeil)<br>><br>><br>><br>> NOTICE: This email message is for the sole use of the intended recipient(s)<br>> and may contain confidential and privileged information. Any unauthorized<br>> review, use, disclosure or distribution is prohibited. If you are not the<br>> intended recipient, please contact the sender by reply email and destroy all<br>> copies of the original message.<br>><br>><br>><br>> _______________________________________________<br>> cisco-voip mailing list<br>> cisco-voip@puck.nether.net<br>> https://puck.nether.net/mailman/listinfo/cisco-voip<br>><br></div></div></body></html>