Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer
> It does seem like a bug. However, you have a complicated dialplan with a lot > of pieces happening at > once so it may not actually be an Asterisk bug but a problem with your > dialplan. To unravel this is > going to take some bookkeeping on your part. Hi Richard, Thanks for the detailed response. Need to get my head around it a bit! I’m going to try to set up a test rig with a less complex dialplan. I’ll then run some tests and will be able to supply sample files if the problem persists. I’m just a little confused what the different is which transfers of inbound queued calls and transfers of inbound Dialled calls. I wonder if it would make a difference if the queue members were Local/DialThisEndpoint_200, instead of PJSIP/endpoint_200. Since the queue dials the members and the member channels inherit the variables correctly, that would mean that Local/DialThisEndpoint_200 would inherit the DYNAMIC_FEATURES. The Local channel would dial the endpoint, and when the endpoint performs a transfer and loses its variables, the Local channel, as the parent, would still have its variables set and the feature codes would still work. -- _ -- 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
Re: [asterisk-users] Disable asterisk ssl how to
On Wed, Aug 08, 2018 at 04:30:52PM -0400, Saint Michael wrote: > I am trying to install Asterisk 11 on debian 9, ... Did you install libssl-dev (Version 1.1) or libssl1.0-dev ? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- 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
[asterisk-users] pjsip: TOS not working any more
Hello! I'm using TOS as shown below with pjsip 13.22.0-rc1 (same with 13.21.1). Unfortunately, the TOS isn't set in reality any more (it used to work some time ago). Transport: == Transport: 0.0.0.0-udp udp 0184 0.0.0.0:5061 ParameterName : ParameterValue == allow_reload : true async_operations : 1 bind : 0.0.0.0:5061 ca_list_file : ca_list_path : cert_file : cipher : cos: 0 domain : external_media_address : external_signaling_address : external_signaling_port: 0 local_net : 192.168.13.0/255.255.255.0 local_net : 192.168.17.0/255.255.255.0 method : unspecified password : priv_key_file : protocol : udp require_client_cert: No symmetric_transport: false tos: 184 verify_client : No verify_server : No websocket_write_timeout: 100 Is there some kernel support needed to get it working? I reduced the kernel to a minimum of modules. Thanks, Michael -- _ -- 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
