Archives : April-2020
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert2. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asteriskThe release of Certified Asterisk 16.8-ce..
I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client from Asterisk to the server. The server sends Server Certificate, Server Key Exchange and Server Done. Than Asterisk sends back a Al..
The Asterisk Development Team would like to announce the release of Asterisk 17.4.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 17.4.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.10.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.10.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.33.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asteriskThe release of Certified Asterisk 16.8-ce..
1) Is there any reason why max_pseudo_channels defaults to 512? I want to increase it by default but at the same time dont want to outsmart the developers if they had a good reason for it.2) I had a look at http://lists.digium.com/pipermail/asterisk-users/2014-March/282607.h..
All,I hope someone can give me a hint.We try to reload the asterisk dialplan config using ansible command module. Using this we just trigger asterisk -rx dialplan reloadNow we want ansibe to fail if there is something wrong in the dialplan.If we ..
Im using an Asterisk 17 dialplan that currently includes:1. many DB gets calls (ie statements like Set(FOO=${DB(Foo/Bar)})2. and a couple of DB puts (ie statements like Set(DB(Foo/Bar)=Foo) or DB_DELETE(Foo/bar))I would like to add an HTTP Provisionn..
Has somebody get combination Asterisk (Im using version 13.32.0, webrtc and iOS (version 13.4.1) with Safari or any other browser working properly in confbridge conference calls? I hope my Asterisk webrtc related settings are not totally wrong, beca..