Archives : March-2018
Unfortunately, upgraded to Asterisk 13.20.0 and we are still seeing strange results in the AMI AsyncAGIExec Result string.First one for the call is successful.Later during the same call, it has characters that would be from some ExternalIVR work ..
i would like to ask how to connect 2 systems. I would like to have an asterisk where it will have all the connections to the outside world (sip trunks) and it will called the gateway. This asterisk will have extension numbers of 3XX. In the LAN th..
–_000_HE1PR0601MB2617D60566ACCEA25B18A62A89A90HE1PR0601MB2617_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableI would really like your help in finding tools for proactive call analytics.Right now i am evaluat..
We are communicating with Asterisk via AMI.Running Asterisk version 13.18.5 on an Ubuntu box.If you look at the event response, the Result field is filled with random characters.Im not sure what to do because that is a completely random result.It ma..
Hi. in my system i have a conference room where someone can call it eg 698dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN.I would like to have a call file and call all participants eg 610-..
Hey List I sometimes use our asterisk server to do some debugging or other PBX and SBC. Now we have a case where a PBX is replying an incomming invite with 180 ringing immediately. It looks like the SBC does not accept this. According to my understand..
The Asterisk Development Team would like to announce the release of Asterisk 15.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 15.3.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 13.20.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.20.0 resolves several issues repor..
I am using AMI Originate to perform a new outbound call.The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header.For the AMI Originate, I pass the caller id information data in the Calle..
Im working on an Asterisk active/passive cluster where the following applies:- members are both VM- /etc/asterisk files are copied from one provisonning server to both VM- asterisk is running on active member- asterisk is not running on passive memb..