[Asterisk-Dev] T.38 pass-through / T.38 support
lov2cod at gmail.com
Fri Jun 3 15:03:30 MST 2005
I am new to Asterisk so maybe I am not making sense, but it is
possible that Asterisk will just reinvite the 2 sides of the call and
not see the RTP stream at all?
On 6/3/05, Matthew Boehm <mboehm at cytelcom.com> wrote:
> Steve Underwood wrote:
> > They must be bypassing * then. * has no way to pass-through a UDPTL
> > stream right now. I don't see how it would even bypass a T.38 stream,
> > though. It lacks the ability to negotiate one.
> I know you find it hard to believe Steve, but I guarantee you this is what
> Caller on PSTN dials ATA fax. Call is routed to AS5300 over PRI. Call
> becomes T38. 5300 sends call to Asterisk and on to ATA via SIP. The warning
> messages about UDPTL t38 that I said previously display during the transfer
> of the fax. Call completes and fax looks great.
> Call originates from ATA. Asterisk sends call to 5300. 5300 says its T38.
> The 5300 will, depending on if local or LD call, terminate either PRI to
> PSTN or to carrier via H323. Call completes and fax looks great.
> All of it goes thru asterisk at some point.
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> To UNSUBSCRIBE or update options visit:
More information about the asterisk-dev