Issue adding analog stations to CM 1

[Zed071]

Good morning!

I've got an Aura 7.0 system (CM/SM/AACC) that has a G450 attached to it for codecs, RAN files, and now an MM716 analog card. We're migrating away (slowly) from a Blue CS1K and the two are connected via SIP trunks, as well as SIP trunks for external calls on Red. I'm having some trouble configuring the analog parts of this setup.

I've made two test sets to try to get these working. I've configured the user in CM, added the DNs to the dial patterns, and gave it a high COS. It's basically an analog version of a maintenance phone for permission levels.

Now when I make some test calls on the analog set, I can call other Red extensions just fine. But when I try to go outside of Red - either to Blue or to the PSTN, I get the Avaya error tone. The same calls from the 9608 SIP set at the desk work perfectly.

When I look at a station trace, I see both calls working on the same route patterns going over the same trunk-groups, yet the analog calls fail with the following results (real numbers and domains have been edited)....

10:22:46 TRACE STARTED 05/14/2019 CM Release String cold-00.0.441.0-23169
10:22:54 active station 3475 cid 0x371a
10:22:59 dial 91705347 routeREFIX|FNPA|ARS
10:22:59 term trunk-group 501 cid 0x371a
10:23:01 dial 917055551212 routeREFIX|FNPA|ARS
10:23:01 route-pattern 3 preference 1 location 1/ALL cid 0x371a
10:23:01 seize trunk-group 501 member 42 cid 0x371a
10:23:01 Calling Number & Name 3475 Test 2, Analo
10:23:01 SIP>INVITE sip:17055551212@sipt.oxxxxxxe.ca SIP/2.0
10:23:01 Call-ID: c7d342ae765341e988d30c299c447b
10:23:01 Setup digits 17055551212
10:23:01 Calling Number & Name +3475 Test 2, Analo
10:23:01 SIP<SIP/2.0 100 Trying
10:23:01 Call-ID: c7d342ae765341e988d30c299c447b
10:23:01 Proceed trunk-group 501 member 42 cid 0x371a
10:23:01 SIP<SIP/2.0 404 Not Found (No route available)
10:23:01 Call-ID: c7d342ae765341e988d30c299c447b
10:23:01 SIP>ACK sip:17055551212@sipt.oxxxxxxe.ca SIP/2.0
10:23:01 Call-ID: c7d342ae765341e988d30c299c447b
10:23:01 denial event 1166: Unassigned number D1=0x9fcd D2=0x1
10:23:01 idle trunk-group 501 member 42 cid 0x371a
10:23:01 dial 917053474373 routeREFIX|FNPA|ARS
10:23:01 route-pattern 3 preference 1 location 1/ALL cid 0x371a
10:23:01 seize trunk-group 501 member 43 cid 0x371a
10:23:01 Calling Number & Name +3475 Test 2, Analo
10:23:01 SIP>INVITE sip:17055551212@sipt.oxxxxxxe.ca SIP/2.0
10:23:01 Call-ID: c7d9d7e765341e988d70c299c447b
10:23:01 Setup digits 17055551212
10:23:01 Calling Number & Name +3475 Test 2, Analo
10:23:01 SIP<SIP/2.0 100 Trying
10:23:01 Call-ID: c7d9d7e765341e988d70c299c447b

I'm thinking there's a setting - possibly in the SM where I have to register the g450 as an entity for CM to allow the calls to pass to either the Blue network or the PSTN network.

Does anyone have a suggestion on where to look next to get the analog lines working?

THanks

John

 

[ZeroZeroOne]

CM appears to be set up correctly since the call finds a route pattern and hits the trunk group.
The "no route available" and "unassigned number" messages in the trace indicate Session Manager doesn't know what to do with the call.
Take a look in Session Manager and confirm you have Dial Patterns that match your dial string and destination. Verify everything under Elements / Routing (Domain, SIP Entity, Entity Link, Dial Pattern, Routing Policies).

 

[Zed071]

Thanks ZeroZeroOne!

I'm taking a look at that now and the G450 wasn't configured in SM.

One possibly dumb question though is what port should be used? I see things like the MAS, the CM, and CS1K are all on port 5060 or 5061 with TLS as opposed to the Audiocodes on 5070 with TCP.

Thanks again!

 

[ZeroZeroOne]

The G450 doesn't need to be configured in SM, just in CM. SM and CM speak directly to each other via CM's IP address (PROCR). Think of the G450 as simply a part of CM and not a separate system or server.
SM only speaks SIP but CM needs a SIP trunk. In your case that's trunk group 501 (based on that trace). Look at the signaling group for that trunk and you'll find TCP/TLS and the port number.
In SM, your Entity Link needs to match what you find in the Signaling Group.

TCP typically uses 5060 and TLS uses 5061. It is possible to set different ports but I would not recommend that unless you know exactly what you're doing.

Verify your Entity Link for CM but I'd focus on Dial Patterns and Routing Policies to make sure SM knows how to process calls. You can loosely think of Dial Patterns as "ARS analysis", Routing Policies as "Route Patterns" and Entity Links as "Trunk Groups" if you are more familiar with CM than Session Manager.

So SM will look at the dial string "17055551212@sipt.oxxxxxxe.ca" and try to find a match in the Dial Patterns. The Dial Pattern could be "1" for 11-digits, "1705" for 11-digits, or even more specific. It could also specify the domain name and even the Location. If a match is found then SM will look at the assigned Routing Policies and send the call to the appropriate SIP entity via the Entity Link. If SM cannot match the String, Domain, and Location, then the call will fail. Most often you'll simply want to assign "all" Domains and Locations - at least for testing, then start limiting as necessary.

There definitely is a learning curve here so I'd read up on the on-line Help available in Session Manager.

I hope that helps.

 

[Zed071]

Thanks again for the direction - it got me thinking what was needed.

I confirmed all the entity links, and revisited the routing policy along with adding in an <ALL> expression into the dial patterns, and from there,magic happened and everything at the main site is working.

Now to go and make lightning strike twice at the client's other site.

JW