vicidial.org

Okay, I have a client that we recently upgraded from 2.0.2 to 2.0.3. Ever since the upgrade their agents have been complaining of echo, however nobody else have been having echo issues. When I did the upgrade I installed asterisk 1.2.23 but recently downgraded them to 1.2.17. They were using 1.2.14 before that. I kept all their original asterisk settings from before the upgrade. The echo only happens when running Vicidial or astguiclient. The agents are using X-lite sip phones. The sip phones all have echo cancellation enabled. When they call from a soft phone without using vicidial or astguiclient there is no echo what so ever.

They have 15 agents and a few managers. The box is a duel core opteron with two gigs of ram. The load average stays around .3 at any given time. This is the output from zttest:

Code: Select all

--- Results after 229 passes ---
Best: 100.000000 -- Worst: 99.804688 -- Average: 99.989922


They have a Sangoma card so I really do not think the problem can be accurate timing.

This is where things get really weird though. This is the output from show channels (the X's are hiding their clients number):

Code: Select all

pdial:/usr/src/zaptel # asterisk -rx "show channels"
Channel              Location             State   Application(Data)
Zap/26-1             s@pbx:1              Up      Bridged Call(Zap/11-1)
Zap/11-1             2302@pri:1           Up      Dial(Zap/g2/2302||o)
Zap/2-1              s@pri:1              Ringing AppDial((Outgoing Line))
Local/91XXXXXXXXXX@d XXXXXXXXXX@dialer:2  Ring    Dial(Zap/g1/1XXXXXXXXXX||o)
Local/91XXXXXXXXXX@d s@dialer:1           Down    (None)
Zap/6-1              s@pri:1              Ringing AppDial((Outgoing Line))
Local/91XXXXXXXXXX@d XXXXXXXXXX@dialer:2  Ring    Dial(Zap/g1/1XXXXXXXXXX||o)
Local/91XXXXXXXXXX@d s@dialer:1           Down    (None)
Zap/5-1              8600052@dialer:1     Up      MeetMe(8600052)
Zap/9-1              s@pri:1              Up      Bridged Call(Zap/47-1)
Zap/47-1             XXXXXXXXXX@pbx:1     Up      Dial(Zap/g1/XXXXXXXXXX||o)
Zap/8-1              8600054@dialer:1     Up      MeetMe(8600054)
Zap/3-1              8600056@dialer:1     Up      MeetMe(8600056)
Zap/4-1              8600051@dialer:1     Up      MeetMe(8600051)
Zap/28-1             s@pbx:1              Up      Bridged Call(Zap/1-1)
Zap/1-1              2166@pri:1           Up      Dial(Zap/g2/2166||o)
Zap/27-1             s@pbx:1              Up      Bridged Call(Zap/7-1)
Zap/7-1              2600@pri:1           Up      Dial(Zap/g2/2600||o)
Zap/25-1             s@pbx:1              Up      Bridged Call(Zap/10-1)
Zap/10-1             2334@pri:1           Up      Dial(Zap/g2/2334||o)
SIP/2310-008058d0    8600054@dialer:1     Up      MeetMe(8600054)
SIP/2302-007ea7d0    8600057@dialer:1     Up      MeetMe(8600057)
Zap/pseudo-160527548 s@pbx:1              Rsrvd   (None)
SIP/2342-007f5200    8600056@dialer:1     Up      MeetMe(8600056)
Zap/pseudo-202979693 s@pbx:1              Rsrvd   (None)
SIP/2316-00780480    8600055@dialer:1     Up      MeetMe(8600055)
Zap/pseudo-180844378 s@pbx:1              Rsrvd   (None)
SIP/2227-007c4540    8600053@dialer:1     Up      MeetMe(8600053)
SIP/2160-00859a20    8600052@dialer:1     Up      MeetMe(8600052)
SIP/2360-007de780    8600051@dialer:1     Up      MeetMe(8600051)
SIP/2288-007a9c00    8600052@dialer:1     Up      MeetMe(8600052)
Zap/pseudo-161642728 s@pbx:1              Rsrvd   (None)
SIP/2288-00792d30    8600052@dialer:1     Up      MeetMe(8600052)
SIP/2334-007b3400    8600054@dialer:1     Up      MeetMe(8600054)
SIP/2360-00821e30    8600057@dialer:1     Up      MeetMe(8600057)
Zap/pseudo-433320167 s@pbx:1              Rsrvd   (None)
SIP/2360-008d06f0    8600057@dialer:1     Up      MeetMe(8600057)
Zap/pseudo-666848866 s@pbx:1              Rsrvd   (None)
SIP/2360-007d60e0    8600051@dialer:1     Up      MeetMe(8600051)
SIP/2360-00769ae0    8600051@dialer:1     Up      MeetMe(8600051)
SIP/2360-007351c0    8600060@dialer:1     Up      MeetMe(8600060)
Zap/pseudo-180600180 s@pbx:1              Rsrvd   (None)
SIP/2360-00834bc0    8600060@dialer:1     Up      MeetMe(8600060)
SIP/2360-0073ea20    8600051@dialer:1     Up      MeetMe(8600051)
Zap/pseudo-124257404 s@pbx:1              Rsrvd   (None)
45 active channels
28 active calls
    -- Remote UNIX connection


If you look SIP/2360 has numerous channels into the same MeetMe as well as channels into other MeetMes. Now I have not really studied the output of show channels in Vicidial religiously but I have never noticed this on other systems. Also at the end of the day when all the agents have logged out and shut off their soft phones those channels will still be in the MeetMes. Is this normal behavior?

Mcargile,

I've noticed the same behavior, it happens when you barge into a meetme room. In my case I (mistakingly) have teached the Manager of the call center to monitor calls dialing 7+ number of session.

I think the right way to go would be to use the LISTEN link inside AST_timeonvad. However right now i have the problem that firefox does not know what to do with a sip address. I will be trying some things today. If I get it to work i will let you know ok?


Greetings,


Alejandro

I have seen that before with SIP ATAs, I'm not sure what caused it either, but in my case I just changed the incominglimit to 1 or two for the offending device in sip.conf and the problem went away.

You could also try turning off call waiting and 3way calling on the SIP device.