Archives : July-2019
All,Is there a way to get Airplay music into asterisk ?I have used Shairport-sync to get Airplay to play audio on my pulseaudio computer – but was wanting that to come into Asterisk ?T..
Using chan_sip, we are able to register with an NEC switch.When I try to REGISTER with PJSIP, the authentication is being rejected.Traces show its using md5 authentication. The packets looks almost identical.The one area that I suspect is causing ..
Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you should be able to dial with SIP credentials in the DP. Is this still possible in recent versions of Asterisk either with chan_sip or pj_sip..
HelloI notice that BLF-buttons on my IP-phone are greyed out and again active after some time. This goes on and on…When looking at Asterisk CLI I see in the SIP NOTIFY :Subscription-State: terminated;reason=timeoutThe BLF-buttons turn on again af..
Hello! Following problem: If there are different trunks (-> different numbers and users / passwords) to the same destination, asterisk (16.x) always uses the same local tcp port for each connection. This is a problem with Deutsche Telekom (they w..
All,I am trying to get the opus codec working with linphone. I followed the instructions… This shows me its loadedcore show translation paths opus— Translation paths SRC Codec opus sample ra..
For easybell, I dont know of any advantage. But thats not very reliable, because Im using easybell for dedicated requirements only. Im considering chan_sip legacy. I wouldnt build any new installation on chan_sip (if there are no technical contradiction..
when using AddQueueMember() to add to a queue, it is possible to add unreachable (non-existing) peers to a queue.Such members show up marked as … (dynamic) (Invalid) … when using the queue show command. Is there a way to disallow adding unreacha..
Is there any way to get shairport-sync audio into aster..
all – I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example:opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex(..