If you place a direct-SIP call from the internet into our system (for example: sip:email@example.com ) the call connects, but it has no audio (it was reported broken on the Teleconferencing page on wiki.mozilla.org)
The "nat=yes" option had disappeared from the [general] section of sip.conf. Since these calls are handled by the "default" (nothing else matches) context, the general/default settings apply. Without this option, it will fail to apply detected IP sources that different from the source listed inside the SIP packets (which is usually wrong if the end user is behind a NAT)
Assignee: telecom → justdave
Status: NEW → RESOLVED
Last Resolved: 5 years ago
Resolution: --- → FIXED
For the record, I had a user trying this out and he couldn't make it work. We involved #servicedesk and were unable to get it to work either. The conf1234@ weren't mentioned on any of the other pages in the intranet or mana, so we assumed this was long broken and a case of bitrot in the wiki. I realize we should have tried to reach out further. Sorry, but glad this found the right person :)
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