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 13:03:09Date Closed:2008-06-28 15:15:08
Versions:Frequency of
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
  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.


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,


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.