|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|
|Environment:||Linux CentOS 5.7||Attachments:||( 0) issue-18199.patch|
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.
I was migrating my php AGI script from asterisk 18.104.22.168 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 22.214.171.124. 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 126.96.36.199 - I am testing to see if the patch will fix it.