kamrun nahar bina
2010-May-07 04:33 UTC
[asterisk-users] Problem of "Playing 'pbx-transfer'"
Dear all,
We have been using asterisk for 4 years. Now we have got problems which
occurs 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' sound
Sometimes 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: 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
HDD: 3.5 SATA 1TB x 2
version of asterisk: 1.4.23.1
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
Asterisk and the User-Agent is connected through the Internet.
......And Is there any solution to solve this problem? I have
investigated 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?
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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 at 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100510/67751657/attachment.htm