<div dir="ltr"><font face="tahoma">A couple of other things:</font></div>
<div dir="ltr"><font face="tahoma"></font> </div>
<div dir="ltr"><font face="Tahoma" color="#000000">All calls to SiteA exts being with 29XX, 3XXX and 4XXX<br>All Calls to SiteB exts begin with 2XXX (other than 9) and 5XXX what would it <br>be possible to route calls to SiteB via a Trunk that has SiteB MTPs and to <br>
siteA via a Trunk that has SiteA MTPs?</font></div><font face="Tahoma" color="#000000"><font face="tahoma"></font>
<div dir="ltr"><br>Thanks<br>Paul</div></font><br><br>
<div><span class="gmail_quote">On 3/28/08, <b class="gmail_sendername">Paul Dillon</b> <<a href="mailto:pdillon@gmail.com">pdillon@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Hi There,</span></div>
<div style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">I am battling with the situation below. I hope the details make sense. If any one has any suggestions or leads or has tried something similar please let me know.</span></div>
<div style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">With Thanks</span></div>
<div style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Paul</span></div>
<div style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"></span> </div>
<div style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"></span> </div>
<div style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><b><span style="FONT-SIZE: 8pt"></span></b> </div>
<div style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><b><span style="FONT-SIZE: 8pt">MTP Required or Not on SIP Trunk to CUPS SIP Proxy</span></b><b><span style="FONT-SIZE: 8pt">?</span></b></div>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><b><span style="FONT-SIZE: 8pt"> </span></b></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Call manager 6.0.1</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Currently we have<span> </span>IOS SW MTPs & IOS HW Transcoders in SiteA (2 routers) and in SiteB (1</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Router)</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">SiteA phones are in the SiteA Device Pool and <span> </span>SiteA <span> </span>MRGL and will use SiteA MTPs</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">SiteB phones are in the SiteB DP and SiteB MRGL and will use SiteB MTPs</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">SiteA DP calls to SiteB DP are G729 and vice versa.</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">We have a SIP Trunk into CUPS SIP Proxy and on to CVP Call Server (SIP Proxy and CVP local to SiteA). Currently MTP required is selected on Trunk. </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">SIP Trunk into CCM is in TrunkC DP and SiteA MRGL and will use SiteA<span> </span>MTPs (Trunk C Device Pool has region settings of G711 to both SiteA and SiteB device pool regions)</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">This is ok for SiteA ingressing calls through the SIP Trunk. However SiteB calls also ingress to CCM through the same SIP Trunk and are routed to IP phones in SiteB and will hence use</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">SiteA DSPs (MTPs) instead of SiteB DSPs (MTPs).</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Typically all calls to SiteA extensions through the SIP trunk will have originated from the PSTN incoming through an IOS gateway in SiteA and entering CVP system and to SIP trunk on routing to an agent</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Typically all calls to SiteB extensions through the SIP trunk will have originated from the PSTN incoming through an IOS gateway in SiteB and entering CVP system and to SIP trunk on routing to an agent</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Internal IP phone users to also need to be able to call to CVP via the SIP Trunk</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">So in the event of MTPs being used across the WAN, what is the bandwidth utilization</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">involved?</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">So if we have MTP required unchecked we run into the following issues:</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">1. Ext calls at G729 through the SIP trunk to CVP. The call is after playing prompts etc in CVP vxml rerouted back to an agent extension G729 through SIP Trunk. The call disconnects</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">2. Ext calls at G729 through the SIP trunk to CVP. Caller hears the prompts back as fuzzy noise</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">3. PSTN caller ingresses to CCM via SIP Trunk and routed at G729 to ext. Ext places call on hold and cant unhold</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">4. Same as call 3 above except call to ext is G711 but the result is the same</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">5. PSTN caller ingresses to CCM via H323 gateway at G711 to ext. Ext answers the call and transfers it at g711 through the SIP trunk and the call</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">fails after a few seconds</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt"> </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">Basically I am firstly wondering can the 5 call issues be resolved without MTP required on SIP Trunk. </span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">If not can I force SiteB calls to use the siteB MTPs and not traverse the WAN for MTPs (despite the SIP</span></p>
<p style="MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: normal"><span style="FONT-SIZE: 8pt">trunk using the MRGL which contains the SiteA MTPs) and what is the bandwidth impact of using MTP resources over the WAN</span></p></blockquote>
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