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 <bina...@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 <al...@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 <bina...@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 5bd1acee539e699b4f5e79c94a348...@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 >>> >>> >>> -- >>> >>> _____________________________________________________________________ >>> -- 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 >
-- _____________________________________________________________________ -- 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