Archives : September-2020
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Were holding ourselves back from moving to PJSIP as we dont appear to have figured out how to force codec preference in a dial plan. The PJSIP Advanced Codec Negotiation document (https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiati..
We don’t normally announce DPMA releases on here but 3.5.5 was just released which resolves a compatibility issue between the latest versions of Asterisk (using PJSIP 2.10) and DPMA. For TCP or TLS traffic it was possible for a crash to occur. Itâ..
–_000_EFCDF2C6785A7B478B3A77A6E7C36369027AB9ECDEmailxaccelnet_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableall,Anyone know an easy way to have the Directory Application lookup all the voicemail contexts in ..
is it possible to call an IP camera?Im thinking about something like bridging with a music stream, but instead of streaming audio, bridge with the video stream from the camera. It would be very cool if I could just call the camera and see whats go..
We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial inv..
Hi. (Asterisk 16.2.1) Im using AMI Originate to initiate calls, and Im passing some additional data in to the dialplan context using the Variable: parameter.Works fine. https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_Originate ..
I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a drop in call. It does not have a certain time, it is random. The audio is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowover..
16.13.0, pjsip Id like to get an alert if a call fails to authenticate: if Failed to authenticate then mail someone the source ip endif As I look at ami or ari, they deal with calls in channels. Is there a way to get failed invites or registers ? s..
I found a problem with latest Asterisk 16.13.0 and SNOM3xx phones (EOL) running the above FW. Incoming calls are no more working, we get error 404 despite the fact that broken registrar is on for the account. Previous FW for this phones dont have ..
All,I built a system which allows people to call a phone number and listen to various online media streams (train yards, radio stations etc). I use ffmpeg + MusicOnHold to play the streams. The system also allows callers to hear pre recorded content.Norma..