On Wed, 2004-08-04 at 14:56, David Gurr wrote: > Solved my own problem ... thought I'd record it here for any others who come > across it. > > The problem arises since Asterisk is trying to get out of the way of the > media stream, by doing a SIP re-INVITE to get the two ends of the > conversation to talk directly. This won't work, as Asterisk is telling the > calling party that the IP address to talk to is the private IP address of > the softphone on the internal network. Adding "canreinvite=no" to the > softphone's stanza in sip.conf solves the problem. > > It would be helpful if Asterisk noticed that it's about to tell the other > end to use a private IP address ... the ranges are well known, and Asterisk > could do an implicit "canreinvite=no" in this situation.
What if both phones are on the private net? I'm sure something is being worked on. > The same problem didn't occur on outgoing calls as the Dial string includes > a "t" for timeout - as per the wiki, this means that Asterisk must stay in > the stream to be able to implement this. t and T are for transfer, not timeout, case denotes which end can transfer. -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ 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
