Archives : October-2019
Since yesterday I have a stuck channel on my Asterisk server and I do not know how to eliminate it: Message/ast_msg_queu macro-dial-one s 59 Up Dial PJSIP/1218/sip:1218@19..
all,No audio flows after holdIm getting this below error after unholding the call.chan_sip.c:10088 process_sdp: Declining non-primary audio stream:audio 14670 UDP/TLS/RTP/SAVPF 107 103 104 9 0 8 106 105 13 110 112 113 101is it a codec issue or someth..
what is the best way to implement email notification on missed call ? So far, I have been using a solution that I hacked together. I create a hangup-handler, inside which I check for ${DIALSTATUS} and use System(): System(/bin/echo … | /usr/bin/m..
FOSDEM – Real Time Communications devroom CfP ============================================= Overview ——– [FOSDEM](https://fosdem.org) is one of the worlds premier meetings of free software developers, with over five thousand people attending e..
The Asterisk Development Team would like to announce the release of Asterisk 17.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 17.0.0 resolves several issues repor..
I have Asterisk 16.2 on Debian. In the Asterisk CLI, I would like to change 2 things: 1) change the keybindings for commandline editing (what in bash is called readline editing of the command line) The CLI is missing some very useful keybindings, ..
We have a product that uses Asterisk via AMI.I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesnt seem..
Im currently using Asterisk 16.4.0 cert version and working on webrtc. For configuration perspective, Im pretty much done with it but here the real issue Im currently facing i.e. when setting parameters max_audio_streams &max_video_streams to any posit..
I am struggling with DTMF detection in Asterisk 16.3. With the Asterisk read dialplan command I get excellent detection. With the AGI GET DATA function the DTMF detection is however often bad. Is the underlying DTMF detection code the same in both functio..
Up to now I have been using one remote server for both incoming and outgoing.The SIP entry looks like this: [combined] disallow=all allow=ulaw allow=gsm allow=ilbc dtmfmode=rfc2833 host=206.380.260.100 defaultuser=6477957868 secret=xxxxxxxxxxxxxx insecure=invite,p..