Fabiano Carlos Heringer
2010-May-26 03:05 UTC
[asterisk-users] Getting "ghost" transfer or music on hold
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font face="Tahoma">Hi Everybody,<br> <br> I´m getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ... In some calls, i get an atxfer or musiconhold in the middle of call, or listening another call (like a cross line) without any intervention of the user. I got this error in about 3-10% of the calls, on a randomic times, and not pattern observed, just happens, and about 5-10 seconds the problem goes out.<br> <br> I can´t identify nothing that can reproduce the error... It´s happens using between SIP calls, or using external interface (Digital Trunk). <br> <br> Got Ideas?<br> <br> Thanks!!<i><br> </i></font> </body> </html>
Prince Singh
2010-May-26 05:26 UTC
[asterisk-users] Getting "ghost" transfer or music on hold
Are your extensions(who get the music between the calls) on SIP ? When the issue occurs, note 1. the SIP peer account with which it is occurring 2. Without hanging up, do a "core show channels" to see how many channels are present for that same SIP peer. If your are unable to identify this yourself, then mail the output of "core show channels" as a reply to this mail. The "core show channels" should be done WITHOUT hanging up the problematic extension -- Regards, Prince Singh Drishti-Soft Solutions Pvt Ltd W: http://www.drishti-soft.com B: http://blog.drishti-soft.com On Wed, May 26, 2010 at 8:35 AM, Fabiano Carlos Heringer < bigu at grupoheringer.com.br> wrote:> Hi Everybody, > > I?m getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ... In > some calls, i get an atxfer or musiconhold in the middle of call, or > listening another call (like a cross line) without any intervention of the > user. I got this error in about 3-10% of the calls, on a randomic times, > and not pattern observed, just happens, and about 5-10 seconds the problem > goes out. > > I can?t identify nothing that can reproduce the error... It?s happens using > between SIP calls, or using external interface (Digital Trunk). > > Got Ideas? > > Thanks!!* > * > > -- > _____________________________________________________________________ > -- 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/20100526/5456ea95/attachment.htm