I am trying to build a hunt group that will queue a call without using an agent license. We have EAS agents but not enough licenses for all groups to be built with EAS (and no interest in purchasing more licenses. When I create a new UCD-MIA Hunt Group, I can set Queue to Yes and there is a spot...
Look into Phybridge PoLRE switches. You can put an IP phone 1200 feet away from the Phybridge switch on single pair of CAT3. That is the spec. We tested to 1600 successfully. That will let you use any IP phone you want. Multiple expansion modules reduces the range. I believe the smallest unit is...
We use the bridged appearance like Kyle555’s suggestion but use the public unknown table to give agents the ability to give location specific caller ID. The same extension with extra call appearances bridged to multiple phones to reduce the license requirement. You could just have a generic...
Centurylink calls the feature 2 B Channel Call Transfer. Other carriers likely have a similar feature. It does cost extra but for us it was the better solution. It our case we prepend digits to the dial string to initiate the carrier transfer. I will add Avaya documentation when I am back in the...
Stations get the Survivable Gatekeeper from station form (page 1) so you would need to bring them up on the core first and then move them to to remotè site. They will retain the settings for awhile. I have seen refurbed phones that still remembered the old extension so you should have plenty of...
I received a minor POW-SUP alarm with error type 2305 on one of the power supplies in a G650. The error count was 3. The alarm self cleared within a minute. A check of the Maintenance Alarm manual gives the description "aux sig lead failure" but there is no further information. An online search...
CDR has the capability to track which trunk a call goes out on if it is routed through CM and configured appropriately. I can't speak to how SIP station work but would think it would be the same. Do you currently have a CDR solution?
Have you tested the DS1 board to see what tests fail? You should see a protocol mismatch there. Sorry, I don't know the tests by heart. If all is good there that an MST trace on the D channel of the PRI would show you all the signaling going back and forth. This is a good resource for the...
Jct22's post is spot on if you are on newer CM and You are talking about an h.323 phone. We have a 5.2 CM and the change needs to be made ip-network-map. You will probably need break out the address range unless you want to move all the phones in that subnet. If it is a SIP phone it needs to be...
Likely you have already Googled the cause code and found that it is an incompatible or unsupported message. Have you run an MST trace to see what Q.931 message is being sent before the cause code 98 is returned?
I would like to add some questions... What about an uptime comparison? We have to reboot our third-party media gateways whenever a routing change is made. Avaya runs on Red Hat Enterprise Linux and doesn't need a quarterly reboot like a Windows server. The annual interchange is only a...
Keep in mind you may loose some features if your 3645's are H.323. SIP phones from another manufacturer only support the 19 features defined in RFC 3261 when used on Session Manager.
You will need the APV-63 adapter to give you the ability to answer from the headset. No settings changes are required in CM. There is an setting in the phone menu for whether you want alerting when offhook and onhook (default), hookswitch only, or neither (if memory serves, you can also set...
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