I had tried this previously and the call simply didn't connect. Freeswitch responded with a 503, and there wasn't much in the way of logging that was helpful to debug the issue. I'm sure I'm missing something obvious here. I'll see if I can get any more info on it, though. Best, Colin On Mon, Oct 2, 2017 at 4:43 PM, Daniel Greenwald <dig1234@gmail.com> wrote:
I think you are looking for: sip_invite_route_uri
On Mon, Sep 25, 2017 at 11:10 AM, Colin Morelli <colin.morelli@gmail.com> wrote:
Hey all,
Trying to figure out how to get the equivalent behavior of fs_path, but only for a single transaction. In other words, I want to start a SIP request hitting a particular proxy, and then simply let Record-Route headers determine where the call should be routed after that.
I might be completely overthinking this, but everything I've tried so far (sip_route_uri, fs_path), have resulted in Freeswitch continuing to use the given route for all subsequent requests, rather than simply falling back to the session route.
Am I missing something?
Best, Colin
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