> 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. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
