[asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?

Rajkumar S rajkumars at gmail.com
Fri Mar 21 08:32:53 CDT 2008


Thanks Atis,

On Tue, Mar 18, 2008 at 3:50 AM, Atis Lezdins <atis at iq-labs.net> wrote:
>  As for current problem - i suspect that device state don't get updated
>  correctly for Queue application, so Queue tries to dial device, and
>  call-limit blocks it from doing so. There's a patch, currently in
>  testing (issue 12127), it should fix this, however if you intend to
>  keep incominglimit to 1, and don't use local channels - there's
>  nothing to worry about.

I had gone through bug 12127. Currently  I am testing with 1.4 Trunk,
dated 20th. so the 12127 patch is applied.

But even in trunk the behavior does not change. I still get the
 [Mar 21 18:18:59] ERROR[29689]: chan_sip.c:3266 update_call_counter:
Call to peer '2501' rejected due to usage limit of 1

But some times, usually when I start testing, I get this new message,
when a call is picked up by agent.

[Mar 21 18:18:28] WARNING[29684]: app_queue.c:3002 try_calling: The
device state of this queue member, Agent/2503, is still 'Not in Use'
when it probably should not be! Please check UPGRADE.txt for correct
configuration settings.

I had gone through the UPGRADE.txt and now my sip.conf is like the following:

[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
limitonpeer = yes

[2501]
type=friend
username=2501
secret=2501
canreinvite=no
host=dynamic
dtmfmode=rfc2833
context = sip
disallow=all
allow=ulaw
call-limit=1
nat=1

Also the queue show command shows that the agent is Not in use, though
the call is being taken.

Agent/2503 (dynamic) (Not in use) has taken 3 calls (last was 26 secs ago)

sip show inuse command shows the following output for SIP/2501 (the
phone of Agent/2503)

asterisk:/etc/asterisk# asterisk -rx "sip show inuse" | grep 2501
2501                      0               1
2501                      1/0             1

To me it seems asterisk (or my configurations) is still not
recognising the fact that SIP peers are busy when attending calls from
queues.

Thanks in advance for any assistance in resolving this,

raj



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