<div dir="ltr">I don't see any SDL alerts but now I have other devices that are registered to the sub that can't call to devices on the Pub. </div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Apr 10, 2013 at 10:57 AM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">If devices are registered to the sub and the call failed routing to the pub I'd guess the pub lost its SDL connection to the sub, or for some reason isn't aware of the devices registered to the sub.<div>
<br><div>
<span style>-Ryan</span>
</div>
<br><div><div>On Apr 10, 2013, at 11:28 AM, Erick Wellnitz <<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>> wrote:</div><br><div dir="ltr">I guess the next question would be: Why did it work before with the dial peer with preference 1 pointing at the publisher which is the secondary in the CM Group and why did it suddenly stop working?</div>
<div class="gmail_extra">
<br><br><div class="gmail_quote">On Wed, Apr 10, 2013 at 10:02 AM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br><blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
<div style="word-wrap:break-word">In 8.x and later trunks have an option to run on all nodes (like RLs). This would likely have saved you some pain.<div><br><div>
<span>-Ryan</span>
</div>
<br><div><div>On Apr 10, 2013, at 10:38 AM, Erick Wellnitz <<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>> wrote:</div><br><div dir="ltr"><div>Here is what fixed the problem:</div>
<div> </div><div>CM Group contains:</div><div>Sub</div><div>Pub</div><div> </div><div>Dial Peers had preference:</div><div>Pub</div><div>Sub</div><div> </div><div>I switched the dial peer preference to:</div>
<div>Sub</div><div>Pub</div><div> </div><div>Now all calls work as expected. I have seen this before but I seem to forget this trunk behavior in troubleshooting.</div><div> </div><div>Thanks for all of the advice! </div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Apr 9, 2013 at 2:36 PM, Tom Piscitell (tpiscite) <span dir="ltr"><<a href="mailto:tpiscite@cisco.com" target="_blank">tpiscite@cisco.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">Sorry, I forgot an important piece of the puzzle. CUCM will only accept SIP messages for a given SIP Trunk on CUCM nodes that are in the CM Group of the SIP Trunk's Device Pool. For example:<br>
<br>
Here are your 3 CUCM nodes in the cluster:<br>
publisher<br>
subscriber-a<br>
subscriber-b<br>
<br>
and you create a SIP Trunk, call it "To_UK_CUBE" and put it in a Device Pool with the following CM Group:<br>
<br>
subscriber-a<br>
subscriber-b<br>
<br>
CUCM will reply with a 503 if you send a SIP INVITE to the publisher from the UK CUBE.<br>
<br>
HTH,<br>
-Tom<br>
<br>
On Apr 9, 2013, at 1:11 PM, Erick Wellnitz <<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>> wrote:<br>
<br>
> Yes, that warning message is present.<br>
><br>
> Here is the sanitized message:<br>
><br>
> 04/09/2013 03:16:28.118 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.252.xxx.xxx on port 46625 index 1864<br>
><br>
> SIP/2.0 503 Service Unavailable<br>
><br>
> Via: SIP/2.0/TCP 10.252.xxx.xxx:5060;branch=z9hG4bKAFEE1E33<br>
><br>
> From: <a>sip:+4420319#####@10.252.XXX.XXX;tag=E8A08E44-2528</a><br>
><br>
> To: <<a>sip:+131260XXXXX@172.16.XXX.XXX</a>>;tag=1141366627<br>
><br>
> Date: Tue, 09 Apr 2013 08:16:28 GMT<br>
><br>
> Call-ID: 9C9AF490-<a href="mailto:A02411E2-96059282-DB3C4A98@10.252.XXX.XXX" target="_blank">A02411E2-96059282-DB3C4A98@10.252.XXX.XXX</a><br>
><br>
> CSeq: 101 INVITE<br>
><br>
> Allow-Events: presence<br>
><br>
> Warning: 399 ccmsub "Unable to find a device handler for the request received on port 5060 from 10.252.XXX.XXX"<br>
><br>
> Content-Length: 0<br>
><br>
><br>
> My trunk points to the correct address and uses port 5060 in both directions.<br>
><br>
><br>
><br>
> On Tue, Apr 9, 2013 at 11:22 AM, Tom Piscitell (tpiscite) <<a href="mailto:tpiscite@cisco.com" target="_blank">tpiscite@cisco.com</a>> wrote:<br>
> Is there a Warning header in the 503? CUCM sends a 503 with the following header in it when it cannot match the IP/port of the incoming SIP message to a SIP Trunk.<br>
><br>
> Warning: 399 <CUCM-Server> "Unable to find a device handler for the<br>
> request received on port <port> from <far-end-ip>"<br>
><br>
> HTH,<br>
> -Tom<br>
><br>
> On Apr 9, 2013, at 11:59 AM, Erick Wellnitz <<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>> wrote:<br>
><br>
> > We expect + followed by 11 digits, which we receive. We rely on the inbound trunk settings to strip it to 4 digits. Would traces show if it is being stripped to 4 digits?<br>
> ><br>
> > RTMT isn't giving any clues either.<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > On Tue, Apr 9, 2013 at 10:52 AM, Kenneth Hayes <<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>> wrote:<br>
> > Have you ran RTMT? Also on the trunk have you checked what you are expecting inbound?<br>
> ><br>
> > Sent from my iPhone<br>
> ><br>
> > On Apr 9, 2013, at 11:49 AM, Erick Wellnitz <<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>> wrote:<br>
> ><br>
> >> There is no carrier involved.<br>
> >><br>
> >> The call is UK CUCM -> UK CUBE -> Global Network -> US CUBE -> US CUCM<br>
> >><br>
> >> All of the branches are part of the whole but operate independently which is why it is set up like this.<br>
> >><br>
> >> The US CUCM is sending the 503 to the UK caller. The only thing I can think of is that for some reason the trunk isn't acknowledging the set significant digits.<br>
> >><br>
> >><br>
> >> On Tue, Apr 9, 2013 at 10:26 AM, Kenneth Hayes <<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>> wrote:<br>
> >> 503 is a carrier issue with the service.<br>
> >><br>
> >> Sent from my iPhone<br>
> >><br>
> >> On Apr 9, 2013, at 11:05 AM, Erick Wellnitz <<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>> wrote:<br>
> >><br>
> >>> We got that working but I think we have another issue that I'm not sure how to verify.<br>
> >>><br>
> >>> I didn't change any of the dial-peers pointing to the CUCM yet, I have only modified the one trunk to use port 5070 so it will not interfere with the current trunk for inbound.<br>
> >>><br>
> >>> Now we get a 503 service unavailable for calls from the remote system. Will I need to change my existing dial-peers to xxx.xxx.xxx.xxx:5060 before I add any dial-peers for use on the new trunk that uses 5070? All inbound worked fine until I added this second trunk.<br>
> >>><br>
> >>><br>
> >>> On Tue, Apr 9, 2013 at 9:28 AM, Nick Matthews <<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>> wrote:<br>
> >>> I've seen this configuration before - sometimes certain call flows require MTP for devices/integrations on the CUCM side. You don't have to set up multiple listening ports on CUBE, as it's indifferent. On CUCM the two trunks have different source ports. If it's outbound from CUCM only that matters, no CUBE changes are required. If you need inbound calls to have MTP only for certain calls changing the port is the best way.<br>
> >>><br>
> >>> I believe with SIP trunks CUCM will not allow two trunks to the same address with the same source port, but it will allow you with H.323 gateways. However, the choice of which of those gateways is basically random and you may think there are two but it's only using one, but that's a H.323 caveat.<br>
> >>><br>
> >>> -nick<br>
> >>><br>
> >>><br>
> >>> On Thu, Apr 4, 2013 at 1:02 PM, Erick Wellnitz <<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>> wrote:<br>
> >>> Doh. I forgot about changing the port.<br>
> >>><br>
> >>> I think that answers my question. I can have one trunk on 5060 for site to site calls w/ MTP and one trunk to 5061 for least cost routing from/to other sites.<br>
> >>><br>
> >>> This seems like it will suit our needs perfectly.<br>
> >>><br>
> >>><br>
> >>><br>
> >>> On Thu, Apr 4, 2013 at 11:09 AM, Anthony Holloway <<a href="mailto:avholloway%2Bcisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>> wrote:<br>
> >>> What's your ultimate goal that you want/need two trunks to a single CUBE?<br>
> >>><br>
> >>> So in CUBE you would do something like this:<br>
> >>><br>
> >>> dial-peer voice 1 voip<br>
> >>> description Trunk 1<br>
> >>> session protocol sipv2<br>
> >>> destination-pattern 1...$<br>
> >>> session-target ipv4:<a href="http://10.1.1.1:5061/" target="_blank">10.1.1.1:5061</a><br>
> >>> !<br>
> >>> dial-peer voice 2 voip<br>
> >>> description Trunk 2<br>
> >>> session protocol sipv2<br>
> >>> destination-pattern 2...$<br>
> >>> session-target ipv4:<a href="http://10.1.1.1:5062/" target="_blank">10.1.1.1:5062</a><br>
> >>> !<br>
> >>><br>
> >>> And then in CUCM you would need to create two new SIP Trunk Security Profiles (Found under System > Security in 8.6), specifying the port in which CUCM should expect to receive the messages. Create your two trunks pointing to the CUBE, using respective SIP Trunk security profiles, and that's how you force an inbound trunk.<br>
> >>><br>
> >>> As for the MTP question: You can require MTP for all calls, which can be good and bad. That's no different from H323 trunks to gateways. The require only when needed comes in to play for SIP Early Offer only. And that's a matter of the calling device and whether or not CUCM receives its capabilities or has to make something up using an MTP's capbilities. DTMF relay mismatch (Out of band versus In band) is a different story, and there's no check box for that. That's simply a function of the Media Manager and the MRGL on the SIP trunk, which will correct DTMF mismatches automatically by dynamically using an MTP as needed. So, three different things going on there.<br>
> >>><br>
> >>> I hope that helped explain it a bit more. Maybe someone else will fill in some of my gaps.<br>
> >>><br>
> >>><br>
> >>> On Thu, Apr 4, 2013 at 10:22 AM, Erick Wellnitz <<a href="mailto:ewellnitzvoip@gmail.com" target="_blank">ewellnitzvoip@gmail.com</a>> wrote:<br>
> >>><br>
> >>> If I have two sip trunks from CUCM to CUBE (one which requires MTP and one which does not) how does the CUBE or CUCM know which trunk settings to use for inbound calls to CUCM?<br>
> >>><br>
> >>> Is it best to make all of the inbound settings the same and do all of the translations on the CUBE or CUCM translation patterns instead of setting the significant digits?<br>
> >>><br>
> >>> I'm also remembering someone telling me a while back that if you uncheck the MTP Required that th etrunk will still allocate MTP if needed. Is that correct? It would allow me to only use the one trunk with translations.<br>
> >>><br>
> >>> As always, thanks for the help!<br>
> >>><br>
> >>><br>
> >>><br>
> >>> _______________________________________________<br>
> >>> cisco-voip mailing list<br>
> >>> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
> >>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
> >>><br>
> >>><br>
> >>><br>
> >>><br>
> >>> _______________________________________________<br>
> >>> cisco-voip mailing list<br>
> >>> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
> >>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
> >>><br>
> >>><br>
> >>><br>
> >>> _______________________________________________<br>
> >>> cisco-voip mailing list<br>
> >>> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
> >>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
> >><br>
> ><br>
> > _______________________________________________<br>
> > cisco-voip mailing list<br>
> > <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
> > <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
><br>
><br>
<br>
</blockquote></div><br></div>
_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
</div><br></div></div></blockquote></div><br></div>
</div><br></div></div></blockquote></div><br></div>