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rajkumars at gmail.com
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 Posted: Mon Mar 17, 2008 1:06 am Post subject: [asterisk-users] update_call_counter: Call to peer '2509' re |
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Hi,
I am using asterisk-1.4.15, My sip configs is like
[2501]
type=friend
username=2501
secret=2501
canreinvite=no
host=dynamic
dtmfmode=rfc2833
context = sip
disallow=all
allow=ulaw
incominglimit=1
nat=1
queue.conf is like
[gen-enq]
joinempty = yes
musiconhold = default
strategy = rrmemory
servicelevel = 60
timeout = 60
retry = 5
wrapuptime=5
announce-frequency = 90
announce-holdtime = yes
monitor-format = wav
ringinuse = no
I am using AddQueueMember to add SIP interface to the queue. Each sip
interface is member of multiple queues. Occasionally I get messages
like
[Mar 17 11:33:01] ERROR[9253]: chan_sip.c:3232 update_call_counter:
Call to peer '2505' rejected due to usage limit of 1
[Mar 17 11:33:01] ERROR[9254]: chan_sip.c:3232 update_call_counter:
Call to peer '2509' rejected due to usage limit of 1
[Mar 17 11:33:01] ERROR[9255]: chan_sip.c:3232 update_call_counter:
Call to peer '2502' rejected due to usage limit of 1
[Mar 17 11:33:01] ERROR[9256]: chan_sip.c:3232 update_call_counter:
Call to peer '2506' rejected due to usage limit of 1
in my asterisk console. At this point the mentioned sip phones are
busy. My understanding is that if ringinuse is set to no, queue should
not try and ring phones that are busy, but some how it is trying. How
can I disable this behavior?
With regards,
raj |
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rajkumars at gmail.com
Guest
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| Posted: Mon Mar 17, 2008 10:00 am Post subject: [asterisk-users] update_call_counter: Call to peer '2509' re |
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On Mon, Mar 17, 2008 at 6:30 PM, Grygoriy Dobrovolskyy
<megahohol at gmail.com> wrote:
| Quote: | Forgot to add:
Multiple queues fo sip phone, it is normal that sometimes it is ringed, as
reported busy for 1 queue and free for another. you limitited incoming call
to max 1 ' incominglimit=1' so 
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My understanding was that if a SIP phone is busy, either due to a call
from queue or a call from another sip phone or even making an out
bound call, the queue application would detect that and skip trying
that channel.
Is this assumption wrong ?
raj |
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atis at iq-labs.net
Guest
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| Posted: Mon Mar 17, 2008 5:20 pm Post subject: [asterisk-users] update_call_counter: Call to peer '2509' re |
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On 3/17/08, Rajkumar S <rajkumars at gmail.com> wrote:
| Quote: | On Mon, Mar 17, 2008 at 6:30 PM, Grygoriy Dobrovolskyy
<megahohol at gmail.com> wrote:
| Quote: | Forgot to add:
Multiple queues fo sip phone, it is normal that sometimes it is ringed, as
reported busy for 1 queue and free for another. you limitited incoming call
to max 1 ' incominglimit=1' so
|
My understanding was that if a SIP phone is busy, either due to a call
from queue or a call from another sip phone or even making an out
bound call, the queue application would detect that and skip trying
that channel.
Is this assumption wrong ?
|
If that would be queue, it would have different log entry. This seems,
a result from Dial(SIP/2505,,).
There are two different settings. You can increase call-limit (or
incominglimit) in sip.conf - so devices will be able to take several
simultenous calls. So, even if SIP device has one call (and call-limit
is more than one), device state of SIP device will be "In Use", and
that's where ringinuse parameter of Queue application comes in - if
set to 0, Queue won't ring and you will see a bit different message.
Hope that this explains architecture.
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.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835 |
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rajkumars at gmail.com
Guest
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| Posted: Fri Mar 21, 2008 8:32 am Post subject: [asterisk-users] update_call_counter: Call to peer '2509' re |
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Thanks Atis,
On Tue, Mar 18, 2008 at 3:50 AM, Atis Lezdins <atis at iq-labs.net> wrote:
| Quote: | 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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