[Asterisk-Users] Question on bridged calls

[Rich Adamson]

> > 
> > Correct. However, you can probably guess that most sip/iax providers
> > also use canreinvite=no anyway. 
> 
> Which is annoying to say the least, but understandable.
> 
> > 
> > > 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.
> 
> Yes I did.  Do some just roll their own server using the iax
> libraries?  Or is iax being supported by commercial platforms now?

I know livevoip.com and teliax.com use asterisk with modifications, and
broadvoice.com does not use asterisk (not sure what they are using).
Some use SER to front end asterisk (or another sip system).

I'd have to guess that any itsp that is offering production iax 
service is likely to be using asterisk with modifications. I've not
heard of any that port iax into a non-* system, but they certainly
could be out there.

Those that do use * know there are lots of issues that haven't been
addressed in code (as opposed to those that simply use asterisk as
a corporate pbx), so they likely have a major support issue trying
to keep up with changes/patches without breaking the components that
are business-required functions.

I don't know of any commercial systems that have ported iax into their
system, but then it probably wouldn't be public knowledge anyway 
given the gpl language and digium's capability to privately license
anything they want (including iax).