Reload causes musiconhold to start on active calls

Hello,

I have Trixbox version 2.0. When I reload the configuration musiconhold cuts in on active calls. For example if extension 413 is on a call and the configuration files are reloaded then extension 413 hears musiconhold. The other party that extension was talking to can still hear extension 413 and everything they are saying.

In the log files I see the following:

Nov  5 15:58:46 VERBOSE[7747] logger.c:   == Destroying musiconhold processes
Nov  5 15:58:46 VERBOSE[7747] logger.c:   == Parsing '/etc/asterisk/musiconhold.conf': Nov  5 15:58:46 VERBOSE[7747] logger.c:   == Parsing '/etc/asterisk/musiconhold.conf': Found
Nov  5 15:58:46 VERBOSE[7747] logger.c:   == Parsing '/etc/asterisk/musiconhold_additional.conf': Nov  5 15:58:46 VERBOSE[7747] logger.c:   == Parsing '/etc/asterisk/musiconhold_additional.conf': Found
Nov  5 15:58:46 VERBOSE[7747] logger.c:     -- Started music on hold, class 'default', on SIP/413-0842a608
Nov  5 15:58:46 DEBUG[7747] channel.c: Scheduling timer at 160 sample intervals
Nov  5 15:58:46 VERBOSE[7747] logger.c:     -- Started music on hold, class 'default', on SIP/414-083eb6c0
Nov  5 15:58:46 DEBUG[7747] channel.c: Scheduling timer at 160 sample intervals

In this case its started moh for extensions 413 and 414.

Does anyone have any idea of what might be causing this?

Thanks in advance,

RichY

Never heard that - you should indicate which Asterisk version you are running.

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx

hello,

I have the same problem,

I am running trixbox 2.2.8
Frepbx 3.3.1
asterisk 1.2.24

Thank you.

that behavior is really strange. Have you checked if people over on the Asterisk forums are seeing issues. A reload simply runs the Asterisk reload command (after updating all the conf files). It seems like there may be a weird bug in Asterisk? If you type “reload” from the CLI do you get the same behavior?

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx

I have the same build and can not get what you describe to happen. So more details are nessary to help track it down and/or duplicate it.

Asterisk 1.2.24, freepbx 2.3.1.0 or 2.3.1.1 using what type of connections? Sip / Zap / IAX, etc. Type of phones, etc…

Here are some additional details of my system:

t1 pri to Sangoma A102 - used for incoming calls and some outgoing calls
10 analog lines for outgoing calls to Sangoma A400 12FXO 8 FXS
No SIP Trunks
110 SIP cisco 7960 phones 10 sip sisco 7961 phones 10 sip sisco 7971 phones
All phones have two extensions. FreePbx is running in Device & User mode
Server HP DL 380 G4 dual 3.4Ghz with 3Gb RAM

problem reported on both 7960 and 7970 while on external calls - (not confirmed)

Trixbox 2.2.8
FreePBX 2.3.1.1
asterisk 1.2.24

I