No. Some of my outbound PSTN carriers' gateways are. In this case, the UAC (say, a Polycom handset) sends a call to Asterisk, who sends the call to Opensips for least-cost routing, who decides on a carrier to send it to. When the 180 with SDP makes it back to Asterisk, it gets sent first as an 180 and then a 183 with SDP.
It's not a big issue. This was one of those things where if it were quick to fix it in Opensips, then excellent. If not, nothing lost. - Jeff On 4/14/09 10:33 AM, "Thomas Gelf" <[email protected]> wrote: > Are you sure that your Asterisk is sending 180 replies with SDP? > > Jeff Pyle wrote: >> I'm sure the trouble does lie elsewhere. But, rather than actually fix the >> problem in Asterisk, if there were a few lines of reply_route script that >> could change a 180 to a 183 when an SDP was present, that's much easier and >> just as effective. Although, granted, it doesn't actually fix the problem. > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
