Summary:ASTERISK-05430: MeetMe Conferencing Application Locks Sip Channels, and Stop Functioning
Reporter:tavis (reticent)Labels:
Date Opened:2005-11-02 13:23:18.000-0600Date Closed:2011-06-07 14:03:18
Versions:Frequency of
Environment:Attachments:( 0) ast_meetme_trace.pcap
( 1) asterisk_console_log.txt
( 2) asterisk.strace
( 3) extensions.conf
( 4) extensions.conf2
( 5) sip.conf
Description:I've setup an extension to call the meetme application, when i call that extension it functions as expected, informing me of my conference number and that i'm the only one in the conference however right after join the conference some problems start occuring:

1.  If i call in with another client (both are SIP based), it does not acknowledge the DTMF tones i send to select the conference room, it acts like it never received the DTMF (it plays the "please enter the conference number followed by the pound key" prompt again)
I have verified that the tones are being sent properly, and otherwise work as expected. (before selecting a conference room)

2.  When i hang up the phone Asterisk does not clear the SIP channel in use by that phone.
Before selecting a conference room calls are properly disconnected by Asterisk and removed from the "sip show channels" list.

3.  After the RTP timeout hits (as configured in sip.conf) it prints a message every second that the call has timed out and will be disconnected.  This continues on forever it seems (12 hours in one case)
Before selecting a conference room, if left idle (no RTP is sent from SIP UAC), the SIP session is properly disconnected/terminated after the RTP idle timer hits.

if add the "de" options (dynamic, select an empty conference room)
the first caller hears the meetme prompts and is put into the first conference room, however the second caller hears nothing, looking at the debug output on asterisk shows that meetme was called and nothing else after that

I'm running on linux kernel (vanilla, with grsecurity patches)
Zaptel drivers were compiled with "make linux26"
There is a T100P card in the system and the "zaptel" and "wct1xxp" modules are loaded
I've tried using the ztdummy module in place of wct1xxp with the same results
Asterisk and Zaptel were compiled with gcc 3.3.5 on Debian Sarge
Comments:By: BJ Weschke (bweschke) 2005-11-02 14:45:42.000-0600

reticent: we need the extensions.conf code you're using to reproduce this so we use the same options to try and reproduce it here.

By: tavis (reticent) 2005-11-02 19:22:02.000-0600

There is a small typo in the extensions.conf file, im not able to delete and re-upload it

The extention number was duplicated

By: Clod Patry (junky) 2005-11-02 20:12:39.000-0600

you can upload your new file as extensions.conf2 and i'll delete the old file after.

By: Mark Spencer (markster) 2005-11-02 21:30:30.000-0600

Also please supply the SIP debug from the phone.  Thanks.  This does not sound like a hang, but an issue with the phone.

By: tavis (reticent) 2005-11-03 14:21:19.000-0600

I've attached the following files:

1.  ast_meetme_trace.pcap - Ethereal trace of session
2.  asterisk.strace - strace -fF -s99999999 output of Asterisk during session
3.  asterisk_console_log.txt - Output of asterisk console during session

I've had a couple people on the Asterisk users mailing list mention that they have done some tests with meetme/sip and havn't come across any problems, however in my case i'm certain that the issue is related to asterisk or somthing that is effecting asterisk.  Looking at the strace and asterisk console log shows that the meetme application is executed by asterisk however it does not seem to run, (in strace the initial executing that works show asterisk statting the various audio files it needs for the meetme sessions, any subsequent calls into the meetme application show the logging statment that asterisk is executing the meetme application but then asterisk does not attempt to stat the audio files.

Also, there is the evidence (in asterisk_console_log) that after the initial call into meetme, asterisk cannot disconnect (or otherwise remove from memory) the active sip sessions, as can be seen at the end of the log where asterisk is complaining that the session has timed out and will be disconnected however it is never removed.  Also attempts to manually remove it via the CLI fail as well.

By: tavis (reticent) 2005-11-06 23:19:25.000-0600

The problem seems to be related to the T100P Card, if i remove the wct1xxp module and load ztdummy the bridge works without issue, however there are noticeable artifacts in the audio using ztdummy.

By: BJ Weschke (bweschke) 2005-11-08 11:37:27.000-0600

reticent: since the original report on this bug seems to have been a hardware issue, can we close it? If you've got an ongoing issue with the sound quality produced by using ztdummy, let's open a seperate bug on that. OK?

By: tavis (reticent) 2005-11-08 16:39:52.000-0600

It does look to be a hardware issue, i'll try with a different card

This ticket can be closed

By: BJ Weschke (bweschke) 2005-11-08 17:24:20.000-0600

closed - after consent granted from the original reporter to do so.