[Asterisk-Users] RE: No incoming audio on incoming SIP calls

[Steven Critchfield]

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 <critch at basesys.com>