Hi, I've put the sip output here : https://pastebin.com/W7M4zxHA Thanks
-----Message d'origine----- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Joshua Colp Envoyé : vendredi 9 juin 2017 11:39 À : asterisk-users@lists.digium.com Objet : Re: [asterisk-users] Asterisk 13 attended transfer alcatel On Fri, Jun 9, 2017, at 04:59 AM, Jason TOMLINSON wrote: > Hello, > > Since upgrading from asterisk 11 to asterisk 13 (I have tested up to > the latest 13.16.0 release), we have a problem with attended transfers > to an alcatel pbx in which the call being transferred still has music > on hold even after the transfer has completed. > Is this a known issue? Any new flags that need setting, etc? There's no filed issues about it that come to mind and no new flags that need setting. I'd suggest providing console output and SIP traffic. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
