I have an ARI application that is registered for Stasis in the dialplan. One of the events I reap in my application is a ChannelDtmfReceived. The thing is, Asterisk 13.6.0 sends me two DTMF for each DTMF pressed (have tried both SIP phones and landline..
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..
This question was asked by Chirag on March 4 2015 earlier, but I am following exactly the same procedure here and I cannot even get my clients to register on Asterisk.Heres my PJSIP.conf:[transport-tcp]type=transport protocol=tcp bind=0.0.0.0:5061…[endpoint_internal](!)type=endpo..
I am using Asterisk 13.6.0 and was wondering if I can programmatically add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk server using API of some sort.Please do let me know.Than..
I am developing a voicemail application and was wondering if I can redirect a caller to voicemail using ARI. Any help is appreciated.Than..
I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually reach the PBX, but for some reason, they are not caught by any of my extensions context.He..
I am trying to set up a default outbound endpoint for my Asterisk 13.6.0PBX, and per https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip, I do in pjsip.conf:[global]default_outbound_endpoint=SillyEndpoint…[SillyEndpoint]type=endpointetc.Howev..
I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I ..
I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs.I am able..
I am trying to force a registration and unregistration with my SIP trunks, but I see pjsip send unregister, but no register.I.e., I am looking for pjsip send register.Is there any such command? If so, why do I not see it in my CLI?Should I upgrade..