Summary: | ASTERISK-00030: app_queue Dynamic member add, also adding unique call ID | ||
Reporter: | Tjardick van der Kraan (tjardick) | Labels: | |
Date Opened: | 2003-07-31 13:03:09 | Date Closed: | 2008-06-28 15:15:08 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_queue |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | AddQueueMember(MyQueue) add user Sip/sipuser-e74u instead of Sip/sipuser. Same with Zap/1-1 instead of Zap/1 | ||
Comments: | By: Mark Spencer (markster) 2003-08-15 13:30:31 Can you please update your bug report in the form of complete sentences? By: Tjardick van der Kraan (tjardick) 2003-08-15 15:58:24 Sorry for being a bit unclear. i'll explain by an example. I have the following extension item: exten => 651,1,AddQueueMember(virtuin) When i dial this with a Sip phone: Executing AddQueueMember("SIP/tjardick-58f4", "virtuin") in new stack The problem is the fact that ir adds the -58f4, which is i think the unique callid ? so when looking at the queue: chromium*CLI> show queues virtuin has 0 calls (max unlimited) in 'ringall' strategy Members: SIP/tjardick-58f4 Zap/1-1 Agent/1001 No Callers So when a call comes in it will look for SIP/tjardick-58f4 instead of SIP/tjardick. I guess the same with the Zap/1-1 it should officially only be Zap/1. Hope this is a bit more clear if not let me know what else you need/ i can do to make it more clear. Greetings, Tj By: Mark Spencer (markster) 2003-08-16 15:26:56 Fixed in CVS By: Tjardick van der Kraan (tjardick) 2003-08-18 03:12:33 The AddQueueMember works perfectly thnx! One little problem is that the RemoveQueueMember is still having the -code in there so logout doesn't work yet :) I thought it was easier to just reopen this bug then to submit a new as i guess it's direct related and probably just as 'easy' to fix. Thnx so far, Tj By: Mark Spencer (markster) 2003-08-18 08:57:24 Okay same thing added for remove. thanks By: Digium Subversion (svnbot) 2008-06-28 15:15:08 Repository: asterisk Revision: 126222 U team/murf/mtxprof/main/utils.c ------------------------------------------------------------------------ r126222 | murf | 2008-06-28 15:15:03 -0500 (Sat, 28 Jun 2008) | 1 line Another experiment; force a coredump at pthread_create before ASTERISK-30, then another at ASTERISK-34, and compare the thread list; at 38, the thread creation is slow; at 34 they are fast. Compared the two; the only diff is that the sip do_monitor is in the handle_request_invite code; and the CLI thread is created. That's it. Tried running Asterisk w/o the CLI, and still the pthread creates are slow, so that isn't it. ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=126222 |