Archives : June-2018
Hello;I do not know if the following feature is depending on the phone (can be configured on the phone it self) or need to be configured from asterisk itself:Is it possible to configure general SIP Phone to have one button that can be used as following..
Hello;Is it possible to configure one button on the IP Phone (like Polycom or general SIP Phone) to indicate (at the phone display) that the line (the line that is connected for FXO port) is busy or not? If it is not busy, the user can press on the but..
I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so often asterisk crashes and then restarts. I am not seeing any core dumps on the box. The only I thing I see every time is a second before Asterisk crashes there is a AAAA lookup ..
My dialplan looks like this:[from-external]Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT)Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)})Exten => _X.,n,Noop(CALLED NUMBER: ${EXTEN})Exten => _X.,n,Ringing Exten => _X.,n,WaitExten(15)Exten..
I am working on a system where I connect to an external API and based on what it gives me I generate the Asterisk dial plan accordingly. I am thinking about my different options and wanted feedback from others on how to best do it.1) Generate conf fi..
Hello;I need to be able to send and receive voice calls through GSM network, so do I need GSM adaptor that will be connected to FXO port or I can use GSM card that can be connected to PCI or PCI-E slot in the computer and asterisk can see this card thro..
156323040995812Salut, Je voulais te demand..
I’m currently using 32 CentOS 5 but it’s now unsupported. I only have a 32 bit processor and CentOS no longer supports 32 bit so I need to move on. I’ve installed the current version of 32 bit Fedora and I can’t get the latest Dahdi to bui..
I am currently using Asterisk 13.21.1 under Ubuntu (Compiled from source).The Dial-by-name directory option that Im currently using: Directory(sip,sip,eb) That allows for first and last name matching. Ive recently enabled forwarding voicemail with ..
Ive just discovered chan_sips ignoresdpversion setting. Do you use it ?If positive which kinnd of issue could you solve with it ?Be..