[asterisk-dev] Meetme problem (talk detection/opt) in 1.6.1.1
Jared Mauch
jared at puck.nether.net
Fri Jul 10 13:12:34 CDT 2009
Greetings,
I've recently upgraded and seem to have identiifed a problem (or two)
with the way that we try to use meetme rooms:
The first issue is that we no longer see user 'talking' information
in the meetme output, even with talk detection (T) enabled.
To solve this issue, I "enabled" the talker optimization, but this
has the result of showing people that correct user state (ie: talking vs not)
but the audio is not passing the bridge for some reason. The phones
are all Cisco 79xx and show the rtp media counters increasing.
I was wondering if anyone else had seen these issues before I start
digging too deep into the code.
I need the 'talking' information to better identify rogue people
on bridges.
(meetme.conf is effectively empty, the bridges are all dynamically created)
Thanks.
- Jared
voip*CLI> meetme list 4028
User #: 01 1076 redacted Channel: SIP/redacted-09a1b8d8 (unmonitored) 00:03:29
User #: 02 1050 redacted Channel: SIP/redacted-b5729718 (unmonitored) 00:00:11
2 users in that conference.
voip*CLI> core show channels
Channel Location State Application(Data)
SIP/redacted-b5729718 s at macro-stdconf:4 Up MeetMe(4028,cdTMs)
DAHDI/pseudo-3181948 s at default:1 Rsrvd (None)
SIP/redacted-09a1b s at macro-stdconf:4 Up MeetMe(4028,cdTMs)
--
Jared Mauch | pgp key available via finger from jared at puck.nether.net
clue++; | http://puck.nether.net/~jared/ My statements are only mine.
More information about the asterisk-dev
mailing list