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<p class="MsoNormal"><span style="color:#1F497D">I should have also mentioned that using Device Pools and local route groups is not a good fit here… we use extension mobility. Since the EM profiles do not have a device pool, they use the settings from the hotel
phone. That’s why I came up with the option of using a CSS on the line, that configuration will follow the user no matter which phone they log into.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Also, I typo-ed my steering digits below… just image that they are both the same length. 000001 and 000002<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
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<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""> cisco-voip [mailto:cisco-voip-bounces@puck.nether.net]
<b>On Behalf Of </b>Barnett, Nick<br>
<b>Sent:</b> Friday, November 06, 2015 12:17 PM<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> [cisco-voip] solution for multiple sip carriers?<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I’m looking for some advice on how people handle situations with multiple sip carriers. We have a mix throughout our company of DNs ported from 2 different sip providers. If only the first provider could have ported everything, but that
didn’t happen. I’m trying to NOT use a proxy as I don’t want to maintain a giant list of DNs. Also of note is that the DNs are not in ranges…<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I have a single CUCM 10.0 cluster and 2 CUBEs (one at each data center). But since I’m just labbing this up, I’m just dealing with a testbed cube and test cluster.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I have 3 ideas on how to handle this. First was to use a CSS on the line that added sig/steering digits on the front… say… 000001 for carrier 1 and 00002 for carrier 2. Then I could peel them off and send them out to the correct SBCs for
each provider. I don’t like this, it makes it confusing for support staff to see how a call SHOULD exit.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">My other idea may be a pipe dream… I’ve added a URI to the DN, so, let’s say
<a href="mailto:1000@carrier1.example.com">1000@carrier1.example.com</a> and <a href="mailto:1001@carrier2.example.com">
1001@carrier2.example.com</a> on the 2<sup>nd</sup> carrier DN. Then I changed the sip trunk to the CUBE to pass this info to CUBE. I see the correct stuff flowing through, and it works with my standard incoming called number and destination pattern dial peers.
Now, I’m trying to figure out how to write a URI dial peer that will do magic things for me. I need to be able to route based on calling party URI… is that even possible? I haven’t been able to find an example like this, so I’m beginning to grow skeptical.
<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">The 3<sup>rd</sup> idea is to use CUSP, which I don’t want to do…<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Incoming calls is not an issue. The problem arises when making sure that an outbound call placed from a number from carrier1 goes out a sip trunk to carrier1. If it doesn’t go to the same carrier that owns the TN, either the caller id will
be overwritten with the sip trunks BTN, or I have to apply a diversion header and all calls show up as long distance (and no longer roll up into billing codes).<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Is there some other preferred method that people use?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks in advance,<o:p></o:p></p>
<p class="MsoNormal">Nick<o:p></o:p></p>
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