[Asterisk-Users] Question on bridged calls

[Rich Adamson]

> If I connect to a provider using iax, and that provider connects to
> his provider using only sip,  the provider I am connecting to isn't
> going to be able to bridge the call and drop out of the media stream
> correct?

Correct.

> If I'm understanding how bridging works, you lose the ability to have
> the media stream going directly between the two endpoints of the call
> with most of the providers out there if you use iax, unless the
> provider has their own tdm network.

Correct. However, you can probably guess that most sip/iax providers
also use canreinvite=no anyway. Why? Because of the number of customers
that have some sort of inexpensive firewall/nat box that would cause
an audio failure several seconds into a call, driving their support
costs skyhigh. You've been around this list long enough to have seen
a high number of * implementors not even understand that, so how 
would you expect a less-technical itsp customer to understand that on
initial account setup?

> Is this correct or am I completely missing something?

You're also assuming that most itsp's use asterisk, and that is not a
valid assumption.