Archives : September-2020
Hello. I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall@sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall@iptel.org) Calling testA works fine while testB fails with CONGESTION. Adding debug ..
All,Im new to Asterisk and Im trying to manage the calls using the rest API. I want to play a media file after 5 minutes the call started, how can Imove the call to a Stasis application using the HTTP API in order to play the media using play API(https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Channels+REST+API#Asterisk16ChannelsRESTAPI-play)?Than..
echo 3 | sudo tee /proc/sys/vm/drop_caches Asterisk should be able to reserve memory and force it to stay locked in memory, exactly like Mariadb does with memlock=1Would the Asterisk developers consider something ..
I am sending email notification when new voicemail is received, with the voicemail message attached. Therefore, once this email is sent, I dont want to keep the original voicemail message on the asterisk server, as the user does not need to call in..
https://issues.asterisk.org/jira/browse/ASTE..
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..