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<div>Often times set providers Will allow you to mask caller ID however you choose. In cases where they do not, They should be able to remove the restriction If you ask. You may have to sign some forms in order for them to do that, If the provider services
e911.</div>
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<div>A few moons ago when I work for a regional service provider, We had a SBC that had a connection from level 3 And a connection from bandwidth.com. We would prefer the level 3 connection for egress and failover to the bandwidth.com connection when the level
3 connection was down.</div>
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<div>Basically we just sent all egress to the SBC and we let HSRP sort out Whether or not it was going out the primary connection or the failover connection.</div>
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<div style="font-size:9px;color:#575757">Sent from my T-Mobile 4G LTE Device</div>
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-------- Original message --------<br>
From: "Norton, Mike" <mikenorton@pwsd76.ab.ca><br>
Date:11/06/2015 4:26 PM (GMT-05:00) <br>
To: "Barnett, Nick" <Nick.Barnett@countryfinancial.com>,cisco-voip@puck.nether.net
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Subject: Re: [cisco-voip] solution for multiple sip carriers? <br>
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<p class="MsoNormal"><span style="color:#1F497D;mso-fareast-language:EN-US">Have you asked either carrier if they can remove the outbound caller ID restriction? With my carrier (PRI, not SIP, but same issue) it was basically just a matter of asking them.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D;mso-fareast-language:EN-US">-mn<o:p></o:p></span></p>
<p class="MsoNormal"><a name="_MailEndCompose"><span style="color:#1F497D;mso-fareast-language:EN-US"><o:p> </o:p></span></a></p>
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<p class="MsoNormal"><b><span lang="EN-US">From:</span></b><span lang="EN-US"> cisco-voip [mailto:cisco-voip-bounces@puck.nether.net]
<b>On Behalf Of </b>Barnett, Nick<br>
<b>Sent:</b> November-06-15 11:17 AM<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"><span lang="EN-US">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></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">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></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">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></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">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></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">The 3<sup>rd</sup> idea is to use CUSP, which I don’t want to do…<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">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></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Is there some other preferred method that people use?<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Thanks in advance,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Nick<o:p></o:p></span></p>
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