Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer
Dear sir, Thanks for your reply. We have tested in another phone like Bria(2.4.3 buid 50906) with same phenomenon. But we are getting same error "Failed to play transfer sound! " during attended transfer. Is there anything which causes this problem? And we are not facing this problem first time. Before we faced in this problem occasionally. But recently, this problem occurs frequently. Is there any other problem or any other prerequisite for this problem? Or is it the problem of asterisk? How we can overcome this problem ? Please give us solution. Thanks in advance Nahar On Sat, Mar 27, 2010 at 1:33 AM, Alyed wrote: > 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 > > 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 %MEMTIME+ COMMAND >> 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk >> >> Our disk space condition is below: >> FilesystemSize 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 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 >>> 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 -- >>> N
Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer
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 > 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 %MEMTIME+ COMMAND > 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk > > Our disk space condition is below: > FilesystemSize 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 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 >> >>> 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 Aster
Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer
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 %MEMTIME+ COMMAND 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk Our disk space condition is below: FilesystemSize 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 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 > >> 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
Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer
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 > 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
[asterisk-users] "Failed to play transfer sound! " during attended transfer
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
