Category : Asterisk Users
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The Asterisk Development Team would like to announce the release of Asterisk 16.21.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.21.1 resolves an issue repor..
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 ..
The Asterisk Development Team would like to announce the release of Asterisk 18.7.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 18.7.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.21.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.21.0 resolves several issues repor..
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..
I need to call 1 number and that number and bring 3 phones into a confbridge. I tried this:; PHONE CONF – Phone group Confexten => 63,1,Originate(SIP/401,exten,63,join_conf)exten => 63,2,Originate(SIP/402,exten,63,join_conf)exten => 63,3,Originate(SIP/404,exten,63,join_conf)ex..
All – I am playing with SIPML5.I was getting an error about wss…. I fixed that by doing :cat privkey.pem > asterisk.pem cat fullchain.pem >> asterisk.pem with my letsencrypt certificate. and setting tlscertfile=/etc/letsencrypt/live/mypath/asterisk.pem..
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 need check sip headers of incoming calls i have hybrid configuration with chan_sip and chan_pjsip enabled so i need check if incoming call is through chan_sip or chan_pjsip because i cant use i.e. ${PJSIP_HEADER(read,something)} on chan_sip is th..