Archives : August-2020
Gang We migrated our voicemail system from asterisk 13 to 16 a couple of months ago. Right after the migration, we got the complaint that vm-intro is being played when the customer had recorded a own announcement. So I assumed we had replaced that f..
Im attempting to run a test of the ARI recording of audio from the channel.When I send the record command, its failing. curl -v -u asterisk:asterisk -X POST http://locahost:8088/ari/channels/mychanntest.1/record?name=mytest&format=WAV&maxDurationSeconds=300&maxSilenceSeconds=3[08..
AllI have a need to create a VM – install asterisk from source – then move the VM to a physical box. When I do this – I get illegal instruction when starting asterisk. So I recompile on the physical box and then everything is fine. Problem is there ..
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–_000_EFCDF2C6785A7B478B3A77A6E7C36369024F1D5A89mailxaccelnet_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableTo all:No matter what I try, I cannot get the system to wait for the admin to join. It just dumps us..
Im trying to transition from AMI to ARI.Running into a small hiccup when I try to create (originate a call) with the caller id name and numberI can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application.c..
I am having a strange problem with a new provider. We already have a couple SIP trunks working fine. We are trying a new provider but we are having one way audio problems with outgoing calls. Incoming calls do have two way audio, only outgo..
Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a number is being played?Here is a test I am running.I am playing multiple portions (sounds and numbers).curl -v -u asterisk:asterisk -X POST http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,sound:demo-instruct,number:123,number:456,number:78..
I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis.I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, ..
I am using asterisk 13.33.0 and POlycom phone with the latest firmware.The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted – it receives a call, after some time(couple minutes) it no longer receives a ri..