We have a customer using Avaya.Currently, they are using chan_sip.We are working to migrate them to PJSIP. I have not been filled in on the exact scenario.I suspect they have some auto forward feature on the number.Rather than their Avaya transferr..
Author : dcropp
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We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio.Some background.. We are using asterisk 16.6.1We are also currently using AMI/AsyncAGI and ConfBridge to bring the part..
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..
We are using AsyncAGI with AMI.On a customer box running asterisk 16.1.1, we are seeing times where asterisk logs indicate its started the agi:async extension. Event: Newexten… Application: AGIAppData: agi:asyncIts taking 2 or more seconds before..
Im trying to track down a CPU spike we are seeing in a system.We have tracked down the spike to a single CPU and TID using that CPU.Indications are that its asterisk running this TID.Im trying to figure out what asterisk is doing on this thread aro..
We have a customer with a system rejecting calls from Asterisk.Its indicating the ptime is 60, but the system admin is saying they only support 20.They are running asterisk 16.2.1 and using chan_sip Is there a way to specify this with chan_sip?Al..
We are running load capacity tests using Amazon AWS configurations.For the tests, we are basically scaling up calls to a second Asterisk box.First box that is calling the second box plays music on hold for 60 seconds, then hangs up the call. My init..
We have a customer where their VM running Asterisk appears to have crashed.Fortunately, we had some debugging enabled. The asterisk messages file has this… (in notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]….)[08/12 15:30:55.8..
We have a system where two calls are in a ConfBridge with recording.This is Asterisk 16.3.0Channel A seems to work perfectly.Wireshark is showing the RTP to/from working fine and having no jitter/lag issues.This call hears everything from channel B.Chan..
I have done additional testing and I havent been able to figure out why its failing.Since my original testing we now set the realm on the authentication section to match what we receive from NEC.Its of the format abc@xyz.com I have verified the md5_c..