Hi. Because I’m newbe I don´t know how call files work and I need some feedback.
I’m working in an automatic dialer on demand via shell scripts on asterisk 11 and CentOS and one SIP channel based in PSTN line. It´s in test stage. One script makes all call files then put them all in outgoing dir. Only the first call file is made call. Others are dismissed for busy line and then moved to outgoing_done dir. Is there a queue for others? How can I ask for view the queue contents? Or should I resend call files to outgoing dir to complete calls?
Just install Elastix and the call center module and you will have the automatic dialing. Also don’t bother running a dialer over pstn lines with asterisk, it wont work properly.
The remarks about using analogue PSTN lines refer to the lack of guaranteed answer supervision when initiating the call, which, by default, means that Asterisk has to assume the call was answered immediately after dialing finished, the lack of guaranteed disconnect supervision, and the calling party clear principle when the call is ending.
When a called party puts their phone down, the network is sent a CLEAR signal. If they pick up again within a certain amount of time, a RE-ANSWER Is sent and the call continues. Analogue lines cannot signal these events to the caller.
After certain amount of time without a RE-ANSWER, the network sends a RELEASE, which will cause a calling analogue line to get disconnect supervision and the calling central office to release the calling line. Not all networks (or parts of one network) will send disconnect supervision, and there is more than one form, so things have to be configured properly for this to work.
The reason for this delayed release is to allow someone to put one instrument down and pick up another one. In the UK, most but not all central offices have reduced this time to a few seconds.