Configure Asterisk with sip trunk


Is is any special considerations to be taken when using a sip trunk with asterisk compared to separate sip accounts?

What should the outbound extensions use for username/fromname etc? All the same or different (within the given number range specified by the trunk)?

Do I initiate the dial commands differently?

Are there any good tutorials on this?


As such there is no specific documentation offically available. You may have to google it.

Certainly there are variations for configuring SIP extension and SIP trunk. Now a days most of service providers provide you with Asterisk configuration like Voice network, Varphonex etc.