[asterisk-dev] SIP call-limit and Realtime

Atis Lezdins atis at iq-labs.net
Fri Jan 18 10:12:49 CST 2008


I've been trying to get working SIP "in use" indications for usage in
queue, however they are not working with Realtime, unless i enable
"rtcachefriends". Are there any significant problems behind this like
with hints and qualify, or this is a bug?

Without "rtcachefriends" i get nothing in "sip show peers/users/objects/inuse".

Asterisk version - 1.4.17


Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

More information about the asterisk-dev mailing list