Search results

See if you can get logs off the endpoint that is returning the SIP 488. Also grab a SIP trace by going to the fs_cli and "sofia global siptrace on". This will add the SIP packets to your log file and perhaps give more information.

I'm working on this now, simply to see how far I can get with it. If you share the freeswitch database amongst the nodes and enable fs_path then this works. I haven't had time to test all the features but park, conferences, extension dialing (local and remote node) all seem ok as long as the...

One possibility is that the extension you're calling has a set of codecs that doesn't match your PCMU or PCMA (unusual not to include one of these). Another is that the extension is requiring encrypted media (SRTP) which is not being offered.

I'm working with a non-local postgresql database and it is slow due to hundreds of queries being fired off for every change of page in the GUI. I'm not sure if it's reusing connections or reconnecting each time. Does anyone have any optimizations that would help improve this?

DigitalDaz, can you say more about this? This is what I want to build, a load-balanced system with all the config and user data stored in a shared or replicated database. We should be able to make user-to-user, conference calls, VM, etc. all work as long as the data layer is shared and the...

Let's solve it in the forum. That's the point of this site. ThriveLife, could you share your dialplan with us?

Sounds like you need to go to SIP Status and reloadxml, rescan your profiles or restart.