Archives : February-2020
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The Asterisk Development Team would like to announce the release of Asterisk 13.31.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.31.0 resolves several issues repor..
Please point me to samples of popping and wiping hangup handlers. I dont need to use the values returned; I just need to clear any handlers before Ipush a new one.Its not clear at https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers+Specificat..
Does polycom support normal multicast from asterisk as the source?Im getting the impression that it only supports its OWN phone to phone multicast or something. Th..
As mentioned in [1], a common pattern is to let everyone monitor everyone except oneself.How do implement this ?Is there something like this:[alice_list]list_item = full_list list_exclude_item = alice[bob_list]list_item = full_list list_exclude_i..
examining the network traffic with wireshark shows that asterisk does not set any QoS values at all. What do I need to do to make asterisk set QoS values (on CentOS 7)? The wiki says to use vconfig to set QoS values[1].What does the skb-priority n..
when sending IMs from endpoint to endpoint with the MessageSend() application, I can check the MESSAGE_SEND_STATUS and send another message to the sender of the message to notify them that their message was not sent when the status indicates it. T..
–000000000000cbeb58059d4c637c Content-Type: text/plain; charset=UTF-8For those of you who actually process SIP MESSAGE requests…Do you use any of the AMI events generated by the Message/ast_msg_queue channel? We want to change that channel to an inter..
fellows VOIPer, If you want to share with the rest of the VoIP & RTC community some news, interesting or breaking through ideas, or even more, some experience you had in terms of designing, integrating or operating various solutions or platform ba..
I can make calls over a SIP trunk as SIP//numberI am trying to make calls over an extension thought using the same format SIP/4452/number – its not working.person says they can connect a software as extension 4452 and it works just fine.I have my register:regis..
Gang I have not yet managed to find a solution to correctly generate CDRs for this situation: Alice calls Bob. Bob has call forwarding delayed 20s to Charlie. Charlie picks up immediately. exten => bob,1,DBget(cfwdly=CFDLY/${exten}); $cfwdly conta..