<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>Tired, you have a quite a few questions. Some will be a matter of opinion. I've commented inline.<br><br><div> </div>
<div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>First basic question is, can SRST work when the local 'branch' router is either a VOIP GW or a regular transit router?</div>
<div><br><span style="font-style: italic;">Theoretically, yes. If you are not pumping a lot of data traffic through the device, then you should be OK. Please be aware of some "exposures", like SIP being open and if you don't close it, people can connect to your router via SIP and commit toll fraud. I don't know how to fix this or the specifics, so you'll have to search the archives.</span><br><br> </div>
<div>If yes and the router is a transit router will bear bones SRST where PSTN is also directly connected to SRST router, what is best/ norm practice in terms of when users want to make calls to the PSTN when in fallback mode i.e. what number are presented to PSTN and also how are inbound calls treated?<br><br><span style="font-style: italic;">This is completely up to you. You can mark the outbound calls with DID numbers or an auto-attendant number. On the outbound dialpeer you can use the </span><span style="font-weight: bold; font-style: italic;">clid network-number </span><span style="font-style: italic;">command. Inbound calls can be routed any way you wish, either directly to the DID station or to an auto-attendant. You would use the </span><span style="font-weight: bold; font-style: italic;">incoming called-number .</span><span style="font-style: italic;"> and </span><span style="font-weight: bold; font-style: italic;">direct-inward-dial </span><span style="font-style: italic;">commands to help achieve this.</span><br><br></div>
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<div>I am thinking about a scenario where you have a DDI ranges presented at remote PSTN hub sites that covers the remote site running SRST. So from PSTN perspective all numbers with lets say 1xxx would route via the hubs and the SRST site is 12xx. In the scenario where WAN is down at remote, my understanding of SRST is that the same DNs will register will the local SRST router i.e. the 12xx... However, I am assuming the PSTN attached to the SRST router is another provider with a different DDI, does this simply mean I have to do some translations to use PSTN?<br><br><span style="font-style: italic;">I'm not completely understanding what you mean here. But simply put, any PSTN connected to the gateway is eligible for SRST registration as an H323 (or SIP?) device. You'll have to ensure that any digits the PSTN sends down translate into some dialpeer or the callers will get error tone.</span><br></div>
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<div>Also on the same sort of subject ! When integrating sites in a CUCM clusters, trying to get some concrete reasons for not configuring the sites with CME and running them as GWs to the cluster versus registering the sites directly with one of the closest clusters; is there any scientific rules for how far latency wise the cluster should be from end user phones ?<br><br><span style="font-style: italic;">All this information is contained in the SRNDs. Personally, I don't like having to reconfigure CME boxes but like to have the automatic configs that SRST has. That doesn't mean this will suit your environment. Things like distributed call centres might mean additional features that SRST does not provide (yet).</span><br></div>
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<div>thnks in advance</div>
<div>TB</div></blockquote></div><br>
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