Process of elemination. Test with multiple phones, check the codec being used
and make sure the file is there and available.
----- Original Message -----
From: kamrun nahar bina
To: asterisk-users@lists.digium.com
Sent: Friday, May 07, 2010 07:33
Subject: [asterisk-users] Problem of "Playing 'pbx-transfer'"
Dear all,We have been using asterisk for 4 years. Now we have got problems
whichoccurs during the attended transfer.During attended transfer, sometimes we
cannot hear the sound of 'pbx-transfer'.
I cannot understand why this is happening?log is : -- Started music on hold,
class 'default', on SIP/113.34.235.13-b7a3f110-- <SIP/0000185148-092db338>
Playing 'pbx-transfer' (language 'jp')
Although it is showing Playing 'pbx-transfer' (language 'jp'), but it cannot
hear 'pbx-transfer' soundSometimes we can hear the sound of 'pbx-transfer'. is
it the problem of network load or phone-set or something else? Please let me
know. I am using x-lite and snom 300.
Before i tested it for memory load, And found out that it is not a memory
problem.Our system is as like as:The number of User agent is: 1650The number of
Actual registered user agent is: 600Our System configuration is :
IBM X3550CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHzHDD: 3.5 SATA 1TB x 2version
of asterisk: 1.4.23.1our 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%stMem: 4147888k total, 3986540k used, 161348k free, 76852k
buffersSwap: 2031608k total, 56k used, 2031552k free, 3170396k cached
PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND23160 root
15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asteriskOur 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% /boottmpfs 2.0G
0 2.0G 0% /dev/shmAsterisk and the User-Agent is connected through the
Internet.
......And Is there any solution to solve this problem? I haveinvestigated in
several places but I cannot find out the reason?I need this solution very
urgently. Is there any one who can solve this problem?--
_____________________________________________________________________
-- 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
