Avaya IP office migrating to SIP guidance

[bishoptf]

Like the title says I have a small business that I have upgraded to 11.0 version and purchased SIP trunk licenses and looking for best way to migrate the service from our existing analog lines. Currently we have about 4 ananlong circuits that act as trunks with 2 inbound DID's for different parts of the business. What I would like to do is add the SIP trunks while keeping the existing ones working so I can test and make sure everything is working properly before migrating the DID's and going live. I have found this documentation from voip.ms that provides what I think is a good overview -

https://wiki.voip.ms/article/Avaya_...n_Avaya_IP_Office_500_v2_.28Version_11.0.4.29

but my question is when adding additional lines how would I go about testing and ensuring that those lines are being used etc...Probably a dumb question but that is what I wanted to do, add the new lines and get them configured, test and verify etc...then at some point once we feel like things are working port the DID's.

Thanks

 

[IPOTS]

Typically you place the SIP trunk order with your chosen service provider.
They will provider you SIP trunk information to configure your SIP line.
Prior to porting the existing numbers over they use a test number to verify connectivity. They will test incoming and outing calls.
Check for 2 way audio and voice quality.
Once that has been tested and confirmed they will ask if you are ready to port. You authorize he port and it's done.

Note: your existing lines will still be operational until you authorize them to port after testing.

 

[ipohead]

If you are very lucky your provider can give settings specific to IP Office!
You should check the trunks are registered and call progress using the System Status app
Once you have set the Incoming Call Route you can add a tag to confirm which line the call is on (see the help for the Incoming Call Route tab)

 

[bishoptf]

I think what I am going to do is to create some sip lines and have them assign me a did and add it into the mix but then add a short code that will allow me to test outgoing and incoming calls while allowing them to continue to use the existing analog circuits. Once I have tested and feel like its ready to go I can schedule them to port the numbers and remove the analog circuits etc...that is what I am thinking right now.