I have been looking online for a definitive how-to on using Asterisk as a
SIP trunk termination point . I am seeing conflicting messages and
methodologies for doing this.
I am not going to use a commercial vendor for this trunk, it will be used in
testing out various customer scenarios and am looking at Asterisk as one
I see two ways to do from all my research
1.use two asterisk in VMs(or on bare metal) to originate and terminate a
– this would be good if one doesn’t have an IP PBX or control over the
2.use asterisk to terminate one end of the trunk by using it to log in to a
sip trunk and then define the peering.
I know the first one is easy, but the second may be the way to go.
What does asterisk need to supply the remote end of the sip trunk? I realize
that it is based on the remote end, I just haven’t seen any example for the
most popular IP PBXs, like Cisco UC or Avaya CM/SES.
Ids this is all trust, I could see the SIP trunk peer entry and trusted and
maybe bypassing the peer username/password in the definition.
Has anyone done this in a proof of concept lab or in production?