kamrun nahar bina
2010-Mar-26 03:55 UTC
[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 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/4f465bf7/attachment.htm
Alyed
2010-Mar-26 05:32 UTC
[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 <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 > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100325/d1c0bff9/attachment.htm