Archives : May-2021
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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..
using 18.4.0I did a ./configure –with-pjproject-bundled –with-jansson-bundled and I still saw a message about downloading .I have sights that cannot download – how can I configure to just use the bundled and not download anyt..
So installed Asterisk 18. changed the modules.conf to remove the noload chan_sip line and did some testing it seems to run fine in that mode.I configured asterisk 18 with–with-pjproject-bundled–with-jansson-bundledCouple questions:1) Is the older chan_..
-I was using asterisk 13.36.0 and tried to specify a MulticastRTP TTL with Channel: MulticastRTP/basic/239.1.2.3:20480/5where 5 is the ttlThis did not work. So I updated to asterisk 18.4.0 tried the same thing and did not work.I remove the /5 and j..
The Asterisk Development Team would like to announce the release of Asterisk 18.4.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 18.4.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.18.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.18.0 resolves several issues repor..
All.Weve put in a check for Do Not Call before a call goes out. However, we have noticed that we cannot seem to pass a hangup reason for a call.For example, Id like to know that this number is on the DNC so our system does not call them back.Is it possi..
Is anyone aware of any way of getting ControlPlayBack to work with an amazon S3 bucket? I know I can put nginx in the middle but I am trying to avoid that..
Running Asterisk 16.17.0We have an interesting scenario where we see Asterisk CPU usage spike to the point the entire system is maxed out.There is a specific scenario where we have two ConfBridges and they are connected via a local channel.Everyth..
Hello! Ive just playing around some time to get NAT and pjsip running with asterisk 18.3 and 18.4 (w/o any patches added). NAT should be used for connection to the trunk. I wasnt able to get it working, because SDP address rewriting just doesnt w..