Jason TOMLINSON
2017-Jun-09 07:59 UTC
[asterisk-users] Asterisk 13 attended transfer alcatel
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? Thanks Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170609/61001673/attachment.html>
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
How are both machines connected to each other ? Through a SIP trunk ? A TDM one ? 2017-06-09 9:59 GMT+02:00 Jason TOMLINSON <j.tomlinson at isi-com.com>:> 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? > > > > Thanks > > Jason > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170609/7ab31ac3/attachment.html>
Jason TOMLINSON
2017-Jun-20 12:50 UTC
[asterisk-users] Asterisk 13 attended transfer alcatel
Hi, I've put the sip output here : https://pastebin.com/W7M4zxHA Thanks -----Message d'origine----- De?: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] De la part de Joshua Colp Envoy??: vendredi 9 juin 2017 11:39 ??: asterisk-users at 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