Asterisk has to be able to execute and rewrite the file - the call file is updated in place and when the call is considered successful, removed.
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 4:28 PM To: [email protected] Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times Also I should note that we use the 'noatime' attribute on the /var filesystem, would this cause the problem below? On 08/29/2011 05:22 PM, Brandon Phelps wrote: > Here is the contents of the .call file. The file is the same before the > move as after (I did a cat on the file after the move, while the phone > was ringing a second time): > > Channel: Local/5703@ext-main > Callerid: "MyCompany" <8005551234> > Set: TicketNumber=1000000 > Set: CallerID_Num=8005551234 > Set: CALLSTATUS=0 > Context: ext-autodialer > MaxRetries: 0 > WaitTime: 45 > Extension: s > Priority: 1 > > We have tried using a SIP channel as well (as opposed to Local) with the > same results. The s extension of ext-autodialer runs an AGI script which > makes use of those Set: variables. > > I can most easily reproduce the problem by simply not answering the > call. After 2 or 3 rings line 2 on the phone lights up indicating > another call. If I reject the first call and answer the second call, > it's the same script. > > Also during my most recent test the following happened: > > 1. I moved file to /var/spool/asterisk/outgoing > 2. Phone rang on line 1 > 3. I let phone continue to ring > 4. After 3 rings, line 2 started ringing (another call from the same > .call file) > 5. I rejected both calls, sending both to voicemail. > 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. > 7. I let the phone ring until it was automatically moved to voicemail > and finally the .call file was removed. > > > On 08/29/2011 11:00 AM, Danny Nicholas wrote: >> Can you post the .call file (with called number blacked out) before >> call and >> after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 >> should >> be from /v/s/a/o). >> >> -----Original Message----- >> From: [email protected] >> [mailto:[email protected]] On Behalf Of Brandon >> Phelps >> Sent: Monday, August 29, 2011 8:45 AM >> To: [email protected] >> Subject: Re: [asterisk-users] Possible Bug? .call files executing >> multiple >> times >> >> On 08/19/2011 09:14 AM, Brandon Phelps wrote: >>> Hello all, >>> >>> We are setting up an auto-dialer to call customers based on the >>> opening of tickets in our internal ticketing system. Everything is >>> going fine so far except for one snag: >>> >>> To test the system we are implementing I am manually moving .call >>> files into the /var/spool/asterisk/outgoing directory like this: >>> >>> asterisk@dialerdev:~# cp test5703.call /tmp/test.call&& mv >>> /tmp/test.call /var/spool/asterisk/outgoing/ >>> >>> This works great and the call is immediately started, however more >>> often than not (ie. not all the time, but most of the time) after >>> answering the call or rejecting it (sending it to voicemail), another >>> call is performed using the same file. >>> >>> I notice that when a call is initiated the .call file is not removed >>> immediately. Instead, asterisk waits until the call is completed >>> before removing the call file, so it seems like 5-10 seconds into the >>> call since the .call file still exists another call is placed. >>> >>> Any advice on how we can avoid this situation and ensure that only one >>> call is made per .call file? >>> >>> The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. >>> >>> Thanks, >>> >> >> Sorry to bring this back up but I am still having this issue and >> haven't had >> any luck resolving it. It should be noted that the .call files in >> question >> are set to MaxRetries: 0, and simply connect the call to the 's' >> extension >> in a custom context. From there the context is pretty complicated, >> running >> some AGI scripts along with some dealing with user input, basically a >> simple >> IVR. >> >> Any help would be appreciated. >> >> Thanks, >> Brandon >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to >> Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
