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Summary:ASTERISK-15943: RECONNECT fails to work for Conference Call
Reporter:starbug (starbug)Labels:
Date Opened:2010-04-11 22:07:28Date Closed:2011-06-07 14:05:07
Priority:MinorRegression?No
Status:Closed/CompleteComponents:PBX/General
Versions:Frequency of
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Description:Hello,

this is my first post, so please help me put it together right.

We are developing on Asterisks using AMI and have noticed that from time to time RECONNECT command fails to properly work, dropping the call on hold

Here are the steps to reproduce the problem:
1. 3000 (Zoiper) make call to 3001 (3CX)
2. 3001 Answer
3. 3001 Init Conference to 3002  (X-lite)
4. 3002 Answer
5. 3001 Complete Conference
6. 3002 leave from conference
7. 3001 Init Conference to 3002   * Zoiper hangup but CTI Client 3000 still talking
8. 3002 Answer

I am enclosing both working and not-working logs in "additional information" section.

I would really appreciate someone get back to me on this, because I expect a lot more people to be affected by this.



****** ADDITIONAL INFORMATION ******

Log file for invalid InitConference:

Edit: Upload debug logs as attachments. - pabelanger
Comments:By: Paul Belanger (pabelanger) 2010-04-12 08:55:22

Welcome, a few things about your posting.

1. Please do not post logs into the 'Additional Information' field, simply upload them to the issue as an attachment.

2. Your steps to reproduce information is good, but you forgot to mention which technology you are using (SIP, IAX2, DADHI).

3. The log file you have posted is not from Asterisk, it looks to be from http://asterisk-java.org/. You have 2 options, a) seek support from their tracker (asterisk-java.org). or b) send the proper asterisk logs for us to debug (see below).

--
Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description didn't include enough information. You may find it helpful to read "Asterisk Issue Guidelines" http://www.asterisk.org/developers/bug-guidelines. [^] We'd be grateful if you would then provide a more complete description of the problem.

At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf).

Thanks!
By: David Sharon (davidsharon) 2010-04-12 13:18:57

Hi pabelanger,

I am working with StarBug in the same project.

Let me provide you the detailed information regarding this situation:

We have three softphones with (SIP) extensions 3000 (Zoiper), 3001 (3CX) and 3002 (X-Lite).

>> 1. the specific steps or actions you took that caused you to encounter the >> problem:

To implement the conference-related functions (e.g. Conference Init, Conference Complete and Reconnect/Conference Cancel), using AMI, we develop a simple user interface.

Our implementation is based on MeetMe.

As you know, to produce "CONFERENCE INIT", before B initiates a consult call to C, A is redirected to queue.

if B initiates "RECONNECT"/"CANCEL CONFERENCE,

>> 2. the behavior you expected, and

after terminating the channel (B and C), A and B will be bridged.

>> 3. the behavior you actually encountered (in as much detail as possible).

Problem occurs at this point:

When B initiates RECONNECT or CONFERENCE COMPLETE, Asterisk drops A (the SIP extension waiting in the queue). What I mentioned happens randomly. For example, it may happen in your first try, or after several successful tries.

We will also attach CLI output.

Question:
Could Pseudo service that we use for MeetMe cause this?
By: Paul Belanger (pabelanger) 2010-04-28 16:22:06

Please attach debug logs (see below) after you have reproduced this issue.

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
By: Paul Belanger (pabelanger) 2010-05-15 18:58:39

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.

Further information can be found at http://www.asterisk.org/developers/bug-guidelines