Archives : February-2016
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list, I was wondering if it were possible for asterisk to do a voice recognition type IVR?Like you know how most banks and stuff do, where they ask you to say your selection or key it in?If this is possible, how can I set this up? Im using FreePBX 2..
Does anyone on this list use Windstream as a SIP trunk provider?If so, would you mind sharing your peer settings?Im using asterisk 13.7.2 and cant seem to get the inbound working correctly (using registration).Outbound is fine, but they are seeing..
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All,Ive setup a Digium G100 VoIP gateway to replace an internal PCI VoIP card in our Asterisk PBX.When using the VoIP card the callerid entries listed in sip.conf were displayed when calling someone over the PSTN.Now, however, though the gateway it j..
Has anyone created any docker images I might be able to use on EC2 for load testing an asterisk platform?I started an instance this morning and was about to load sipp and other tools, and then thought surely someone must have done this already.Id l..
​Dear friends:Is there a way to execute a macro or sub-routine after we send the invite before we receive anything like a 200 OK, 183, etc..
I want to test a SS7 connection with 3 virtual machines running with Wmware workstation player. I tried all configurations in network setting in Wmware but although I can ping other machines and then see their mac address but it seems TDMoE cannot f..
I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway. I am able to make calls outbound through the gateway, but I am not able to make calls into the PBX from external PSTN.Specifically, an incoming call is _received_ by Asteri..
Tonight community services may have intermittent availability due to maintenance. This maintenance will begin at approximately 8:00 PMCST[1] and should last no longer than three(3) hours, ending around11:00 PM CST.The affected services are:* All Aster..
All, Ive been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not.Ive SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting directmediadeny|directmediapermit to offl..