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
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