[OpenSIPS-Users] 180 w/ SDP -> 183
Jeff Pyle
jpyle at fidelityvoice.com
Tue Apr 14 18:41:10 CEST 2009
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" <thomas at gelf.net> 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.
>
>
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