[asterisk-users] "Failed to play transfer sound! " during attended transfer

[Alyed]

so doesn't looks like overload

Could it be a problem with the firmware of your softphones? Have you been
using some new phones lately? someone else in a different thread pointed on
attended transfer bugs with SNOM phones.

> We are eagerly waiting for your solution.
Hope we can help but don't so much pressure on me or the listers :)

Alyed



2010/3/26 kamrun nahar bina <bina187 at gmail.com>

> Dear sir,
>
> Thanks for your reply.
>
> our memory size is 4GB.
> concurrent calls no : 30.
> Our memory condition is below :
>
> Cpu(s):  0.3%us,  0.7%sy,  0.0%ni, 98.5%id,  0.0%wa,  0.1%hi,  0.3%si,
> 0.0%st
> Mem:   4147888k total,  3986540k used,   161348k free,    76852k buffers
> Swap:  2031608k total,       56k used,  2031552k free,  3170396k cached
>
>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
> 23160 root      15   0  440m 415m 5688 S  4.3 10.3 398:13.93 asterisk
>
> Our disk space condition is below:
> Filesystem            Size  Used Avail Use% Mounted on
> /dev/mapper/VolGroup00-LogVol00
>                       901G  245G  610G  29% /
> /dev/sda1              99M   18M   77M  19% /boot
> tmpfs                 2.0G     0  2.0G   0% /dev/shm
>
>
> We are eagerly waiting for your solution.
>
> Thanks in advance.
>
> Nahar
>
>
>
> On Fri, Mar 26, 2010 at 2:32 PM, Alyed <alyed at vivoxie.com> wrote:
>
>> If you didn't have this problem before I'll check up for any changes
>> lately (i suppose you have done so, but ask this just to be safe)
>> I see you have lots of agents and also lots of hard disk space, so I guess
>> disk space is not an issue. Please check it anyway.
>>
>> how many concurrent calls you have? 2 GB in RAM seems little against 600
>> registered agents.
>>
>> Alyed
>>
>>
>> 2010/3/25 kamrun nahar bina <bina187 at gmail.com>
>>
>>> Dear sir,
>>>
>>> We have been using asterisk for 4 years. Now we have got problems which
>>> occurs during the attended transfer.
>>> But we are not always getting this problem. Sometimes it happens. But now
>>> we cannot understand why this is happening?
>>>
>>> problem is:"Failed to play transfer sound! "
>>>
>>> The log of asterisk is as like as followings:
>>>
>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP message -
>>> rejected , no callid, len 366
>>>
>>>
>>>
>>> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was
>>> pretty quick last time, waiting for them.
>>> [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was
>>> pretty quick last time, waiting for them.
>>>
>>>
>>>
>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on
>>> dialog 5bd1acee539e699b4f5e79c94a348361 at 113.34.235.8
>>>
>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Received bye, issuing owner
>>> hangup
>>> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was
>>> pretty quick last time, waiting for them.
>>> [Mar 25 17:58:40] WARNING[13591] res_features.c: Failed to play transfer
>>>
>>>
>>>
>>> sound!
>>>
>>> Our system is as like as:
>>> The number of User agent is: 1650
>>> The number of Actual registered user agent is: 600
>>>
>>> Our System configuration is :
>>> IBM X3550
>>> CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz
>>>
>>>
>>>
>>> Memory: 2GB
>>> HDD: 3.5 SATA 1TB x 2
>>> version of asterisk: 1.4.23.1
>>>
>>> Asterisk and the User-Agent is connected through the Internet.
>>>
>>>
>>>
>>> ......And Is there any solution to solve this problem? We have investigated in several places but we cannot find out the reason?
>>> We need this solution very urgently. We are eagerly waiting for reply.
>>>
>>> Thanks in advance
>>>
>>>
>>>
>>> Nahar
>>>
>>>
>>> --
>>>
>>> _____________________________________________________________________
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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
>>
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>>   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
>
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