[asterisk-users] Asterisk Queues problem- URGENT

Syed Nasruddin nasruddin at ncel.com.pk
Mon Aug 4 01:34:06 CDT 2008 




Hi,

Can anyone help me on this. I am really stuck.again defining the problem
briefly.:

1. Second New card TDM240P added to machine.
2. Only FXO modules i.e 24 FXO.
3. Asterisk detected all the ports successfully and when I run module
reload chan_zap.so it list allthe FXO ports correctly.
4. when I can on any of the newly added lines there is a clear ring on
the orginators phone while no activity detetcted by asterisk. It just
keep quiet. It looks like call is not being detected by the card to my
asterisk.
5.   4 port FXO card which was previously installed is functioning
properly only this new added card is causing problem.
6. I have 12 new lines and only one of the lines is generating below
mentioned logs in asterisk:

== Starting post polarity CID detection on channel 18
    -- Starting simple switch on 'Zap/18-1'
[Aug  4 11:09:29] NOTICE[12255]: chan_zap.c:6169 ss_thread: Got event 17
(Polarity Reversal)...
[Aug  4 11:09:31] WARNING[12255]: chan_zap.c:6232 ss_thread: CID timed
out waiting for ring. Exiting simple switch
    -- Hungup 'Zap/18-1'
  == Starting post polarity CID detection on channel 17
    -- Starting simple switch on 'Zap/17-1'
[Aug  4 11:09:35] ERROR[12256]: callerid.c:564 callerid_feed: fsk_serie
made mylen < 0 (-1)
[Aug  4 11:09:35] WARNING[12256]: chan_zap.c:6202 ss_thread: CallerID
feed failed: Success
[Aug  4 11:09:35] WARNING[12256]: chan_zap.c:6215 ss_thread: CallerID
returned with error on channel 'Zap/17-1'
[Aug  4 11:09:37] WARNING[12256]: chan_zap.c:6232 ss_thread: CID timed
out waiting for ring. Exiting simple switch
    -- Hungup 'Zap/17-1'


Can anyone decipher this code??? What is happening?? Please give me some
cluess to work on. In my Zapata.conf I have following two lines related
to above logs:

Cidsignalling= v23
Cidstart = polarity


Please help./

Syed nasr (MONDAY 04/08/2008)


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Syed
Nasruddin
Sent: Friday, August 01, 2008 8:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Queues problem



Thanks,

Yes that was the problem I have added joinempty=yes. It is now working,.

Right now another critical problem has come up which I have mentioned in
my previous email. I am copying the problem here again:

was initially running only with one TDM800P card having 4FXO and 4 FXS
port then I later added another 24 port FXO card. So now in total I have
now 32 FXO ports for in coming calls. Card was successfully integerated
and all the ports were detected by asterisk. Just few minutes back the
POT lines were also ready and now I am getting additional errors which I
am pasting here.

 

starting simple switch on 'Zap/17-1'[Aug  1 19:00:26] ERROR[3416]:
callerid.c:564 

callerid_feed: fsk_s erie made mylen < 0 (-1)[Aug  1 19:00:26]
WARNING[3416]: 

chan_zap.c:6202 ss_thread: Caller ID feed failed: Success
[Aug  1 19:00:26] 
WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error
on channel 'Zap/17-1'

[Aug  1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med 
out waiting for ring. Exiting simple switch    -- Hungup 'Zap/17-1'
-- Saved 
useragent "X-Lite release 1002tx stamp 29712" for pee r 1001[Aug  1
19:18:29] 
NOTICE[3162]: chan_zap.c:6678 handle_init_event:  Alarm cleared on
channel 17

  == 
Starting post polarity CID detection on channel 17    -- Starting simple
switch on  'Zap/17-1'

[Aug  1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 

(Alarm)...
[Aug  1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed 
out waiting for ring. Exiting simple switch  Hungup 'Zap/17-1'

Please help on this urgent.
I cant upgrade right now  since I am not confident abt upgrade procedure
and any other problems occuring after that. This is my only production
machine.

thanks

 
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mark
Michelson
Sent: Friday, August 01, 2008 7:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Queues problem

Syed Nasruddin wrote:
>  
> 
> Hi,
> 
>  
> 
> I have Asterisk 1.4.18 and I have been running call center queues on
it. 
> Today it suddenly stopped adding inbound calls to queues. I am facing 
> with following error:                       _app_queue.c:3939 
> queue_exec: unable to join queue "myqueue"_
> 
>  
> 
> In extension file:
> 
>                                           Queue(myqueue|t|||120)
> 
>  
> 
> And my agents are joining in following manner:
> 
>                                            Exten => 
> 1001,1,AgentLogin(SIP/1001)
> 
>                                            Exten => 
> 1000,1,AgentLogin(SIP/1000)
> 
>              
> 
> One more thing my asterisk successfully captures the call , it plays 
> music on hold but when it starts to push the call in queue it gives
out 
> this error.
> 
>  
> 
> Any one help me out. It's a production machine.
> 
>  
> 
> Thanks
> 
>  
> 
> Syed nasr
> 

When diagnosing this sort of issue, it is a good idea to check the value
of 
QUEUESTATUS to see why the caller could not enter the queue.

The most common reason for a caller to not join the queue is because 
joinempty=no is set in queues.conf (if you do not have joinempty set at
all, 
then it defaults to no). This setting causes callers attempting to join
a queue 
to not be able to if the queue is empty or if all the queue members are
paused 
or have an "invalid" device state.

Another possibility is that you have a maximum length set on the queue
and so no 
more callers can join because the queue is full.

My suggestion is to see what the QUEUESTATUS is. If the status is
JOINEMPTY, 
then you can issue a "queue show" command on the CLI to see what the
current 
states of your queue members are. It may be as easy to fix as setting 
joinempty=yes in queues.conf. If the status is something else, though,
then a 
different fix may be in order instead.

Mark Michelson

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