Archives : June-2015
Long story short – I have an ancient Britsh Telecom phone attached to my Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call quality is excellent. However, dialling out is impossible, as Asterisk consistently mis-reads the num..
Im doing an upgrade from Asterisk 11 to 13. Im following the guide at https://wiki.asterisk.org/wiki/display/ast/setting+up+PJSIP+REaltime to setup realtime, as I use realtime on Asterisk 11 too.Im getting the following error when trying to connect ..
I have found several places where it is explained how to do this, and Ihave got the following setup, but it is not working (the provider claims they are not getting a proper caller ID setting from me).I have a number of extensions that are shortha..
I am trying to add header to fax while sending fax. I can add some information to header but some information added automatically (date and page count). I want to format this information.Header => Date | ${LOCALHEADERINFO} | ${FAXOPT(localstationi..
B.H. all.We have noticed many calls on our PBX get stuck – the other end sends BYE, and our side sends ACK but the call remains active (no hangup event on AMI, the call is listed in core show channels) and its impossible to hang up until asterisk..
on 11.17.1Trying to debug vm. as suggested by voip-in..
My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i cant use the email feature thats built into voicemail.What I want to do is execute a remote command with the voicemail as an argument.The remote mach..
Dear asterisk users, We would like to announce 2 VoIP activites organised by XiVO. For those whove never heard about the project: XiVOis a free/open source telephone system under the GPLv3 that has been based on asterisk since 2005. More info on ..
HiCan asterisk handle asterisk variable variables?For example:If I were to..
,I am trying to setup a Asterisk setup in AWS instance CentOS6.5 . Ihave installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws ..