[asterisk-users] Preserve CallerID on transfers
Hi, it´s possible to mantain the original CallerId when making transfers? (atx or blind) Example: A calls to B, A transfer to C, C see the CallerID of B, and not A... It´s possible? Thanks1 -- _ -- 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] Solving the CDR mess of attended transfers
Em 07/09/2010 17:15, Miguel Molina escreveu: El 07/09/10 14:49, Fabiano Carlos Heringer escribió: Is there a way to solve the mess on CDR caused by CDR Transfer? anyway, by paid support, no paid, or another way... Im going crazy about this. My boss is really furious because he don´t understand nothing on the CDR. I tried the 1.6.2.11, Asterisk 1.8 beta, and everything still the same. Any solution? Thanks! Hi Some quick measures: 1. Enable unanswered=yes on cdr.conf and try to see if it helps you with the CDR. 2. Try using CEL (Channel Event Logging) in 1.8-beta and try to see if that helps in a definite way. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Hi, will make this change on my cdr.conf About CEL on asterisk 1.8 i tried some test on my test server, he really logs each event on log, but i did not understood how he will work on a user view (most simple). It´s possible to log this events on a database such mysql? 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
[asterisk-users] Solving the CDR mess of attended transfers
Is there a way to solve the mess on CDR caused by CDR Transfer? anyway, by paid support, no paid, or another way... Im going crazy about this. My boss is really furious because he don´t understand nothing on the CDR. I tried the 1.6.2.11, Asterisk 1.8 beta, and everything still the same. Any solution? 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
Re: [asterisk-users] MOH in the middle of the call
Hi, I have the same problem too, but i can´t provide more information, because i can´t find more information, just in the log show me that´s it´s starting a MOH normally. It´s happen on random way, without nothing similar on each call. Using Elastix 1.6, x64 with ATA LinkSys PAP2. Em 01/09/2010 16:57, Stefan Schmidt escreveu: > Danny Nicholas schrieb: >> *From:* asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dario >> Quiroz >> *Subject:* [asterisk-users] MOH in the middle of the call >> >>> Hi, I have a very strange problem. In the middle of the call the MOH >> starts for 30 seconds approximately. >>> After this the call run normally. >>> Anybody have an ideia or has some similar problem? >>> Thanks in advance!! >> You haven’t provided enough information. Guesses would be that it is a >> normal thing or that you are getting some kind of perhaps SIP error >> that is causing a momentary disconnect, triggering MOH until the >> condition resolves itself. >> > i had some problems like this, but only when a snom phone transfered a > call. if you use asterisk 1.6.x this could also be an answer bug which > is allready been fixed. this bug cause some strange issues with moh and > wrong codec write formats. > > but without further information its just a random guess ;) > > best regards > > steve smith > -- _ -- 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] Getting "ghost" transfer or music on hold
Yes, the both extensions are SIP. The problem to get the "core show channels" output it´s happen too fast, so I can´t get the output at the moment of the call... I have the log of CLI output, with all types log enables (WARNING, NOTICE, DEBUG), but nothing of unusual in the log shows. Prince Singh escreveu: Are your extensions(who get the music between the calls) on SIP ? When the issue occurs, note the SIP peer account with which it is occurring 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 <b...@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 -- _ -- 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] Getting "ghost" transfer or music on hold
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
