Hi.I am using meetme application and I am interested in switching to confbridge, but there seems to be no way to do certain things in the dialplan with confbridge. How would I get the number of users in a particular conference?I want the leader to o..
Hi.I am having a problem with a conference call on my server which a vps in the cloud.I am using chan_sip and meetme.What I get is a bit of a staticy or robotic sound, but it goes away if the user lowers the volume a bit which we can do with *4 in meet..
HelloI notice a major difference in what Asterisk console is telling me (which seems correct) and what Asterisk Manager is telling.A SIP user is called, and the phone does not ring. This is the situation.On Asterisk console I see (which seems to be..
The Asterisk Development Team would like to announce the release of Asterisk 19.2.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 19.2.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 18.10.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 18.10.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.24.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.24.0 resolves several issues repor..
How does an external program get notification of new registrations ?Would that come over the AMI or anything ?T..
Were running asterisk 16 with Realtime.We have queues configured in realtime.The Timeout setting appears to have an upper 2 minute limit. Even when setting the timeout in the queue to 600 seconds, the agent is no longer rung after exactly 120 secon..
i have two asterisk boxes need transfer call from second box to first one pstn -> asterisk1 -> dial(number 555) -> asterisk2 -> TRANSFER (number 444) -> asterisk1 dialplan on asterisk1 (using chan_sip) [some_context] exten => 555,1,Noop() same => n,dial(SIP/asterisk2/5..
So I have CentOS 7 server running asterisk 18.8.0 – all is good.I unplug that server – plug in a ubuntu 20.04 server at the same IP address. let my 3 devices reconnect to the ubuntu server….When I pick up the polycom phone and dial it connects. I h..