Dropped calls - Dropped audio - stuttering ivr - on new FreePBX distro -restore settings from backup

General Help
1

I have a new install of FrePBX distro 2.9.0.7, Asterisk (Ver. 1.8.7.1)on a Dell dual core opteron 2220 server w/2tb raid mirror. It is behaving badly!

Symptoms are: calls drop after attendant picks up queue. Call audio dropping out on one side of a call. dtmf droppouts, IVR audio dropping out and stuttering/repeating (including voice prompts, like during transfers, ) These symptoms are random and not related to call volume.

As for volume:
I have 6 queues about 25 active calls/queues, about 55 total extensions.

Configuration: Other than a stock install, i did yum update and updated the FreePBX core and modules, then restored from a backup of an old server that was going bad, that server had no audio problems though.

Old “server” Dell P4 computer, was originally a elastix, but I had problems with that server and cisco phones so I upgraded asterisk to 1.6 and freepbx to the latest. That server was running ok with asterisk 1.6, but reporting hd errors and the cdb never worked after I upgraded asterisk.

My thought are that the entire problem is related to the internal timing. I have tried adding internal_timing = yes in asterisk.conf but no change.

I could move forward with compiling in different timing sources but last time I recompiled asterisk, I broke as much as I fixed.

I might try building a new server with FreePBX distro or Asterisk Now on a spare dell 2x P4 XEON I have sitting around (but this is scsi and so no 1tb drives… no call recording option)

I could build a new server with some other freepbx that uses asterisk 1.6

I was thinking about paying for support, but the options for support seem vague and possibly costly given the $225/hr rate… Hint support sales - we like per ticket pricing.

Can I check/change the timing source without re-compiling everything?

Any other advice?

2

Update: Same issue, I installed asterisk now with asterisk 1.6.2.11, updated freepbx to the same version and restored a backup.

This seems only to occur when a call reaches the “operator” then the call is traansfered to another desk. I have tried transferring with the phone, with iSymphony, same result. I have changed the operators cisco 7960 frimware from 8.8 to 8.12.

These calls go from -inbound route, time condition, call flow control (holiday greeting), to a queue, then to operator. 1/15 calls, operator has audio problems, Operator transfers to extension, 8/10 calls have audio problems.

I have several hundred calls a day on the same system that go to other queues, with no problem. No problem when a call is transferred to voicemail.

From the settings I can see that the internal timing is still enabled with this distro.

PBX Core settings

Version: 1.6.2.11
Build Options: LOADABLE_MODULES
Maximum calls: Not set
Maximum open file handles: Not set
Verbosity: 0
Debug level: 0
Maximum load average: 0.000000
Minimum free memory: 0 MB
Startup time: 17:54:00
Last reload time: 17:54:00
System: Linux/2.6.18-194.el5 built by root on x86_64 2010 -08-24 20:45:59 UTC
System name:
Entity ID: 00:04:23:cb:b8:d8
Default language: en
Language prefix: Enabled
User name and group: /
Executable includes: Enabled
Transcode via SLIN: Enabled
Internal timing: Enabled
Transmit silence during rec: Enabled
Generic PLC: Enabled

  • Subsystems

    Manager (AMI): Enabled
    Web Manager (AMI/HTTP): Disabled
    Send Manager FullyBooted: Disabled
    Call data records: Enabled
    Realtime Architecture (ARA): Disabled

  • Directories

    Configuration file:
    Configuration directory: /etc/asterisk
    Module directory: /usr/lib/asterisk/modules
    Spool directory: /var/spool/asterisk
    Log directory: /var/log/asterisk

I am running out of ideas with the exception of the internal timing. Thanks in advance for anyone that might be able to help.

3

I have decided to buy a Sangoma UT50 after reading up on similar issues on pbxinaflash forum.
http://pbxinaflash.com/forum/archive/index.php?t-9084.html

4

The UT50 did not fix the problem. It does increase the overall call quality for every call. I have literately tried every thing I can think of. Adjusting QOS settings, Changing the inbound route to go directly to the phone (cisco 7960 with 8-12-00) update the firmware, change the canreinvite, nat. The most strange thing, I have 4 other did’s that come into the same server the same way but go to another queue. All those agents are fine. Outbound calls are fine.

5

Very strange. The operator can put a call in the parking lot, then the user can pick up the parked call with no audio problems.