We have a customer where their switch sends pidf+xml presence information in the SIP INVITE message.Does Asterisk process this pidf+xml information?Does it store this in a channel variable that a dial plan could access?If not, does it store present t..
Author : dcropp
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I have been using Asterisk 18.11.2. Just tried Asterisk 18.12.0 and am running into a problem with the res_pjsip_transport_websocket.Using Ubuntu 20I use a bash shell script to compile Asterisk with settings. I didnt modify any settings from Aster..
We currently use the Queue.Under app_queue, it uses module res_monitor (which is on the to be deprecated list). Is it safe to continue using Queue (app_queue)?DanThis email is intended only for the use of the party to which it is addressed and may cont..
We tell asterisk to use the slin format for ExternalMedia.However, the unicast channel is selecting ulaw formatand the RTP data is indicating its ulaw format.Anyone know why ulaw format would be on chosen?[10/12 16:13:39.396] DEBUG[1665] http.c: H..
When we perform ExternalMedia with the slin format, we are still receiving ulaw rtp packets.Asterisk logs show its selecting ulaw. Im guessing we are missing a menuselect or configuration setting. Anyone have any suggestions for the possible cause ..
We are running Asterisk 16.17.0 and discovered what we think is an issue.We have a single call in a ConfBridge. Tell the ConfBridge to start recording. We see non-stop audiohook.c 160 samples failures.As soon as we stop recording (AMI ConfBridgeStopReco..
Running Asterisk 16.17.0We combine calls into a ConfBridge using AMI with AsynAGI.Executing actions to ConfBridge channels into the same ConfBridge. Its a very large and busy system, so there are dozens of these that may happen during a second (differ..
We have an extremely busy/large customer.They run fine most of the time, but periodically asterisk will output FRACK refcount related messages.It doesnt seem to be related to the volume, because its not breaking during their peak times.When this happe..
I am working with a very large customer running Asterisk with PJSIP.Systems total channels have been over 2500 (which includes hundreds of local channels and ConfBridges) when the issues occur. Its running on a Hyper-V VM with 12 CPU cores. Things w..
Has anyone used Audio Sockets with Amazon Transcribe?Im still very new to the Audio Socket and have only just started looking at Amazons Transcribe documentation so there may be something I am missing. Im looking for live transcription of the call..