Using call Files for massive dialer… I guess if is there a way to gracefully stop call spool processing…I wish i could stop call files execution at certain hours,especially the 2nd or 3rd scheduled temptative set via retry options…
Moving call Files via cron lead me to lost call Files that are in execution state…so prior to move i wish i could gracefully stop them…any idea?
Thanks
the usage of call files to build auto dialer it is not a good option, calls files impose many limitations
I have built my own Auto Dialer, but instead of call files,I prefer AMI Originate Action and PHP. Then using Cron Jobs and Jquery I can schedule the start and stop day for the calls. Also MYSQL to store the dates and numbers
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For this project i’m facing a strange behaviour by a Panasonic pbx wich gave my asterisk box outside sip trunk.
Panasonic tells Asterisk that outside calls are always and immediately answered even if they are ,for Asterisk point of view , in ringing state…So i have to manage call files adding variables to rebuild call files…a mess…so rebuild dialer for now it’s out of question…Customer never knows he could lost at least 0,1% calls…
If you are using analogue lines to the Panasonic, there is going to be no answer supervision, so Asterisk is going to have to assume an immediate answer. I don’t think any analogue PABX provides answer supervision to phones.
Any serious generator of automated calls, or phone spam, would be expected to interface using ISDN, either directly, or via an ITSP that does.
PBX admin said that this is usual Panasonic’s behaviour with Sip Trunk to let external pbx call outside…Panasonic has PRI outbound connectivity…Every call that i do via Asterisk is given answered…
When i said that this would let my system going crazy they reply that’s the way Panasonic do Sip Trunking…
That’s the issue with Panasonic PBX then. They shouldn’t answer the call unless until external party answers the call over PRI.
I know but you know they are monolithic PBX…we are the swiss knife pbx…
So we have to circumnavigate all issues…
I don’t think you can do much in this case. If Asterisk receives 200 OK from other end on SIP then the call is considered answered for Asterisk
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I have made a workaround…to solve this…i’ve always back a 200 OK…
And on the channel i have italian ringing tone…4 seconds silelnce 1 second beeping…
4 seconds are too much time to detect silence on the channel…and colud lead Asterisk to confusion…
So i have recorded a file with pure silence and given as “moh” on the dialing channel…
Whenever a callee says Hallo,backgrounddetect detect human voice and i could set the call as answered…
[call-file-dialer]
exten => 10,1,NoOp()
same => n,Set(MACHINE=0)
same => n,Answer()
same => n,BackgroundDetect(silence,40,450)
same => n,Noop(answering machine detected)
same => n,Set(MACHINE=1)
same => n,Hangup()
exten => talk,1,GotoIf($[${MACHINE}=1]?2:3)
exten => talk,2,Goto(ringtoparked,s,1) ; this is a machine
exten => talk,3,Goto(dialerreply,11,1) ; this is a human
I just would like to ask if you could share such Auto Dialer that I and other people could adjust it to their needs.
I use call files but they are not so efficient.