| Summary: | ASTERISK-18939: CLONE - [regression] LIMIT_CONNECT_FILE does not get played to caller when using dial() app | ||||||
| Reporter: | David O Reilly (trendboy) | Labels: | |||||
| Date Opened: | 2011-11-30 07:37:35.000-0600 | Date Closed: | 2011-11-30 09:17:11.000-0600 | ||||
| Priority: | Minor | Regression? | |||||
| Status: | Closed/Complete | Components: | Applications/app_dial | ||||
| Versions: | 1.8.7.1 | Frequency of Occurrence | Constant | ||||
| Related Issues: |
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| Environment: | Linux CentOS 5.7 | Attachments: | ( 0) issue-18199.patch | ||||
| Description: | Hi guys This problem still exits in the latest version as I just upgraded and broke my asterisk until I reapplied this patch. Should this patch now be made into the next release as a bug fix? it is not a "nice to have" or anything but a working documented part of the product that should be fixed for everybody. The patch works great anyways so its low priority for me but it is there as a problem that should be in the next release as a fix rather than a patch from an old version that doesn't seem to be fixed. Thanks guys for your time. You're all awesome. Dave I was migrating my php AGI script from asterisk 1.6.2.15 to 1.8.5 and everything seemed to be working except when using the dial(,,L(x:y)) app to play a message once the call is ANSWERED. It appears that asterisk is not honoring the channel variable "LIMIT_CONNECT_FILE" as it did in version 1.6.2.15. I also check the other variable "LIMIT_WARNING_FILE" and this on does get played back when 'y' time is remaining. -- AGI Script Executing Application: (Dial) Options: (SIP/provider1/xxxxxxxxxxx,45,rL(60000:30000)) > Limit Data for this call: > timelimit = 60000 ms (60.000 s) > play_warning = 30000 ms (30.000 s) > play_to_caller = yes > play_to_callee = no > warning_freq = 0 ms (0.000 s) > start_sound = mysounds/misc/B-31 > warning_sound = mysounds/misc/B-32 > end_sound = | ||||||
| Comments: | By: Matthew Nicholson (mnicholson) 2011-11-30 09:17:11.103-0600 This patch has already been applied to asterisk 1.8. I am not sure which release it is in or will be in. If it is not fixed in the version you are using, continue to use this patch until a version with the fix is released. By: pedr (pgtpgt) 2011-12-04 11:48:45.960-0600 I still having the same problem, even with patch. Which version of Asterisk this option works well? By: Michael L. Young (elguero) 2011-12-04 16:57:08.046-0600 It looks like it should be in the 1.8.8 release. By: David O Reilly (trendboy) 2011-12-15 14:42:07.360-0600 Just to update this, the bug is still present in Asterisk 1.8.7.2 - I am testing to see if the patch will fix it. | ||||||