Archives : July-2019
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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(..
Have you looked at the actual ARI events that are occurring and the (presumably SIP) signaling for the call when it occurs? — Joshua C. Colp Digium – A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW – Huntsville, AL 35806 – US Ch..
how about sticking in a pbx between [c] and [h]so when [h] hangsup you send to [s] if that is 3rd party else i dont see how you could redirect [c] at allelse maybe ask them to have [h] redirect [c] to [s] then [h] will also be out o..
I am designing a solution for a hotel booking call center with the following(mandatory) design:After the call from the customer with the booking agent is complete (and the Hotel PBX disconnects from the call), a second PBXtakes over to conduct a sur..
Weve recently replaced an old Meridian phone system (Analog) with Asterisk and signed up for Spectrum SIP trunks. The service gets installed on July 8th and I was hoping somebody that may have already gone through the process could give me some hints…
how can I create a self-signed certificate for asterisk which actually works?I had one that did work, and yesterday it suddenly quit working for no reason.I had to spend hours to create another one that would finally work, and it suddenly quit work..
Hello!I have a Polycom phone and sometimes I need to transfer calls without picking them up to local extensions. Polycom has a transfer button which sends SIP 302 packet to asterisk. Another local extension, receiving the call, sees not the origi..