Archives : May-2019
Home » 2019 (paged 20)
Joshua Is there a way in PJSIP to send the audio between the parties always, unless one of the parties is behind a NAT?A session refresh would work. That my only problem with PJSIP. This is routine in the old si..
We are working with an Avaya switch.We send them a REFER.If the transfer is successful, everything is great.If it fails (busy), they send an INVITE in-dialog with a media attribute of inactive.After that, they send a 486 busy. The problem is Avaya basica..
all,Ive got a program that connects via AMI and acts upon the voicemail message waiting event.Id like to be able to force one of those events at will instead of having to wait for the voicemail app to cause the event to get emitted.Is this possible?..
all,Im migrating a box from PJSIP with normal Flatfiles to ODBC/Realtime, Also 16.0.1 to 16.3.0. After adding a few peers to the new RT box, I noticed a delay in call processing. All I had done thus far is added a few endpoints for upstream carrie..
All.I have a question – I have an AGI script that may run for 10seconds, or it may run for 60 seconds while an agent becomes available(agents are geographically dispersed).Is there a way to have the music play in the background while the AGI scri..
We have a scenario where we have ConfBridges assigned to specific users. An administrator wants to listen to the owner of a ConfBridge calls and may want to whisper instructions to them.I thought ChanSpy would be the perfect solution for this.Dur..
Hello!
Im just wondering if its possible to decrypt sips packages in Wireshark while asterisk runs as sips client (connecting to the provider w/
tls 1.2)? I dont use an own certificate.
Thanks M..
According to some older documentation it is possible to disable subscription to any AMI events during the authentication process by adding Events: off as a parameter to the login action.Looking at the Digium documentation from Asterisk 13 onwards: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_Lo..
HelloI am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10).There is local video input (I can see myself), but never video on the receiving side.This is the case in both directions (so it ma..
Can anyone help with an issue regarding the H264 profile level being passed through Asterisk? We have a video call like this:Caller A -> Asterisk -> Called BCaller As INVITE SDP offers profile-level-idB801f, and Called Breplies a 200 OK containing profile-level-idB8..