[asterisk-users] Hold problem with Queue

Mon Apr 4 02:25:28 CDT 2011

Sorry, during the weekend I don't have access to logs.

srvcom*CLI> core show channels
Channel              Location             State   Application(Data)
0 active channels
0 active calls
24 calls processed

=========== SIP/1CDF0F4AD346 call 3600

== Using SIP RTP CoS mark 5
    -- Executing [3600 at interne:1] Set("SIP/1CDF0F4AD346-00000041",
"CALLERID(name)=APPEL DE TEST") in new stack
    -- Executing [3600 at interne:2] Set("SIP/1CDF0F4AD346-00000041",
"CALLERID(num)=NE PAS REPONDRE") in new stack
    -- Executing [3600 at interne:3] Answer("SIP/1CDF0F4AD346-00000041",
"") in new stack
    -- Executing [3600 at interne:4] Queue("SIP/1CDF0F4AD346-00000041",
"TestQueue,,,,180") in new stack
    -- Started music on hold, class 'default', on SIP/1CDF0F4AD346-00000041
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
    -- SIP/1CDF0F4A35F2-00000043 is ringing
[Apr  4 09:01:10] WARNING[8440]: mp3/interface.c:216 decodeMP3: Junk
at the beginning of frame 49443303
    -- SIP/E05FB9818972-00000044 is ringing
    -- SIP/002699ABE031-00000042 is ringing
	-- Nobody picked up in 10000 ms
    -- Nobody picked up in 10000 ms
    -- Nobody picked up in 10000 ms
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
    -- SIP/1CDF0F4A35F2-00000046 is ringing
    -- SIP/E05FB9818972-00000047 is ringing
    -- SIP/002699ABE031-00000045 is ringing
    -- SIP/002699ABE031-00000045 connected line has changed. Saving it
until answer for SIP/1CDF0F4AD346-00000041
    -- SIP/002699ABE031-00000045 answered SIP/1CDF0F4AD346-00000041
    -- Stopped music on hold on SIP/1CDF0F4AD346-00000041
    -- Locally bridging SIP/1CDF0F4AD346-00000041 and SIP/002699ABE031-00000045

srvcom*CLI> core show channels
Channel              Location             State   Application(Data)
SIP/002699ABE031-000 3600 at interne:1       Up      AppQueue((Outgoing Line))
SIP/1CDF0F4AD346-000 3600 at interne:4       Up      Queue(TestQueue,,,,180)
2 active channels
1 active call
25 calls processed

=========== SIP/002699ABE031 Hold SIP/1CDF0F4AD346
=========== SIP/002699ABE031 Hangup

srvcom*CLI> core show channels
Channel              Location             State   Application(Data)
SIP/002699ABE031-000 3600 at interne:1       Up      AppQueue((Outgoing Line))
SIP/1CDF0F4AD346-000 3600 at interne:4       Up      Queue(TestQueue,,,,180)

=========== If SIP/1CDF0F4AD346 want hold SIP/002699ABE031, it's work
    -- Started music on hold, class 'default', on SIP/002699ABE031-00000045

Thanks for your help

2011/4/1 Satish Patel <satish_lx at hotmail.com>:
> We need logs or console output
>
> --
> Sent from my iPhone
>
> On Apr 1, 2011, at 9:01 AM, Elensarde <elensarde at gmail.com> wrote:
>
>> Yes, when the caller are in the queue
>>
>> New informations :
>>
>> - If  A call B directly and B hold A, it's work...
>> - Test with Asterisk 1.8.0, 1.8.1, 1.8.2, same problems...
>> - Phones : Cisco SPA502G / SPA508G / SPA509G
>>
>> 2011/4/1 Satish Patel <satish_lx at hotmail.com>:
>>>
>>> Do you have music on hold configure?
>>>
>>> --
>>> Sent from my iPhone
>>>
>>> On Apr 1, 2011, at 3:39 AM, Elensarde <elensarde at gmail.com> wrote:
>>>
>>>> Hello List,
>>>>
>>>> First, sorry for my bad English skill, I'm French.
>>>>
>>>> We have an asterisk 1.8.3.2 built from sources with a simple Queue :
>>>>
>>>> [TestQueue]
>>>> strategy=ringall
>>>> timeout=15
>>>> retry=1
>>>> timeoutpriority=conf
>>>> ringinuse=yes
>>>> wrapuptime=2
>>>>
>>>> member => SIP/002E31,0,Agent A
>>>> member => SIP/1CA3F2,0,Agent B
>>>> member => SIP/E08972,0,Agent C
>>>>
>>>>
>>>> And this dialplan (extension.ael) :
>>>>
>>>> 3600 => {
>>>>  Answer();
>>>>  Queue(TestQueue,,,,60);
>>>>
>>>>  Playback(invalid);
>>>>  Hangup();
>>>> }
>>>>
>>>>
>>>> When somebody call this exten, an Agent take the call without problems.
>>>> But when he want hold this, phone try to hold the caller without
>>>> success.
>>>> Finally, no signal in the caller-line and the agent-line is hangup
>>>> (for the phone), I not have errors or warnings in logs...
>>>>
>>>> Any ideas ?
>>>>
>>>> Thanks in advance, and kind regards,
>>>>
>>>> Elensarde
>>>>
>>>> --
>>>> _____________________________________________________________________
>
>
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>             http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>              http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>