Conf Bridge: External Callers Muted

Asterisk 13.6.0

Odd situation, but I have a zultys MX250 that is handling traffic to our ITSP provider. I wanted to enable conference calling and (since Zultys pricing is exorbitantly high) I rolled a FreePBX/Asterisk build on Centos 6.7. Set up a few conferences, and also set up IVR/Inbound routes for each conference rooms (in case someone fat fingers the room number). I’ve disabled SELinux and iptables for now to debug this problem.

Occasionally, a user will be muted when dialing in. They can still hear the conference in progress, but no one can hear them. Doing extensive testing, I’ve figured out that:

  • If the user presses 1 to mute, and then 1 to unmute, they’re able to talk
  • It only seems to surface with external calls from what I can see. If someone initiates a conference from a local (Zultys) extension and someone else dials in from a cell phone, maybe 1 out of 15 calls the cell phone user is muted. Vice versa (external cell phone starts the conference, internal extension(s) dial in to the conference), the problem doesn’t occur. Over 50 calls, not a single one muted.
  • Disabled user menus. Problems still persists.

In the conference settings, “Leader Wait/Mute on Join” are not enabled and the “m” flag is not set in MYSQL asterisk.meetme. Debug shows the following:

pbx.c:4818 pbx_substitute_variables_helper_full: Function REGEX(“m” ) result is '0’
pbx.c:4983 pbx_extension_helper: Launching ‘ExecIf’
– Executing [[email protected]:6] ExecIf(“SIP/”, “0?Set(CONFBRIDGE(user,startmuted)=yes)”) in new stack

app_confbridge.c:1040 conf_update_user_mute: User SIP/ is unmuted: user:0 system:0.
app_confbridge.c:1040 conf_update_user_mute: User SIP/ is unmuted: user:0 system:0.

CLI> confbridge list
Conference Bridge Name Users Marked Locked?
================================ ====== ====== ========
885390 2 0 unlocked

I have a full debug 5 dump of a muted call on pastebin, but I guess new users can’t post links. :confused:

Not quite sure what the issue is, but nothing in the debug log seems to indicate that the external call is muted. Any ideas?

Obvious stupid question - external users are outside your local LAN, right?

Any chance they are experience an RTP issue with your firewall.

Yup, external users are outside of our LAN. Layout is as follows:

Level 3 > SRX550 > Zultys > Asterisk physical server

So far, it seems limited to external callers. There currently no firewall rules between Level3 and the SRX.

I’m culling through a level 20 debug/verbose asterisk log and tcpdump to see if there’s any timeouts or glaring problems.

I’ve disabled the menu for a conference room and retested. Was able to reproduce the muted line (from a cell phone), and pressing any key on the number pad allowed to start talking.