Hi,
I’m using SIP phones and limit them to make only 1 outbound call at a time. I set in In sip.conf I set call-limit = 1.
The problem is Asterisk doesn’t send proper signals to the caller:
- busy when busy
- some connection error when phone disconnected
call-limit = 1 limits also inbound calls and when phone is busy for all inbound calls I get an error:
NOTICE[14861] chan_sip.c: Call to peer 'XXX' rejected due to usage limit of 1
VERBOSE[14861] app_dial.c: -- Couldn't call SIP/XXX
VERBOSE[14861] app_dial.c: == Everyone is busy/congested at this time (0:0/0/0)and variable ${DIALSTATUS} = CHANUNAVAIL, not BUSY. Caller gets connection error message instead of simple busy tone.
If I check for ${DIALSTATUS}=CHANUNAVAIL and use Busy(10) command in dialplan I get wrong signal when SIP phone is disconnected (busy instead of connection error).
I thought of checking peer status before Dial() command with ChanIsAvail() but it doesn’t work for me at all.
ChanIsAvail(SIP/XXX) always sets ${AVAILSTATUS} = 0 when peer is available, busy, disconnected or congested (by call-limit).
sip.conf
[XXX]
type = friend
context = from-local
callerid = "Wywolanie" <XXX>
call-limit = 1
username = XXX
secret = gay
dtmfmode = rfc2833
host = dynamic
nat = yes
qualify = yes
qualifyfreq = 60
deny = 0.0.0.0/0
permit = 10.0.0.0/8
Version: Asterisk 1.8.10.0-1digium1~squeeze