I am using asterisk 220.127.116.11 for real time. When an agent is taking call its status still does " SIP/201 (dynamic) (Not in use)". How can i over come this problem.
You need to set a call limit before Asterisk will monitor use levels on SIP channels.
Thx I did it and it working. But it only works when i make user in sip.conf file. But if we make user in realtime of sipbuddies its status does not change… I experimented call limit is not working in sipbuddies table
Now Please guide