Convert incoming pstn to sip trunk


Just wondering if it’s possible to hand off all incoming analog trunk calls to sip calls, then free up the analog trunk.
We have been having issues porting our number…


In Manitoba, Bell MTS Centrex and PRI service can. But, when you dial DTMF, the switching center must be able to hear the DTMF.

However, Bell MTS analog and Shaw (another carrier here) cannot.

You dial a star code, then you hear a message from the switching center telling you to dial the number you want to transfer the call to.

Given the uncertainty of my answer, have you looked into simply call forwarding the lines until the porting is complete?


Yes, I added call forwarding to the line today, will take 48 hours according to bell :frowning:
Thanks for the response though!


Most of our (US based) customers use the forwarding technique. We set up a system, test it, give them temporary SIP numbers that no one ever needs to know except to set up forwarding, then they forward existing numbers to the temporary numbers and off they go. When the ports complete, no one cares because everything is already set up.

In some cases, mainly in hunt group scenarios, they don’t have the ability to dial out on the extra lines, so they typically call the current carrier to set up the forwards. No DTMF needed, but you may wait on hold for a while with customer support.


We forward a lot.
If the analog handoff happens at the provider level then you can have more than one call, if you try to do it at the UCM then you only get the one inbound call.