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