vicidial.org

Hi Guys

Having a strange issue at moment.

When we dial out from vicidial the customer is really quiet and the agent struggles to hear them.

When monitoring using the blind extensions monitoring and listening to call recordings the agent is clear as crystal but the customer is really quiet.

If we dial directly from a softphone, therefore bypassing the vicidial and meetme conferences the customer is fine.

Is there any way of increasing the gain on the customers call from within the conference?

Regards

gain is only used for hardware channels, not voip.

your use of this phrase tells me that you are someone who has played this game before, and therefore dangerous (ie: you know too much ... and likely have been "playin' round in there").

is this installation "stock", or have you modified settings already?

AFAIK, the sip channel is a straight signal, and volume is determined by the recipient of the sound.

So ... check the sound setup of the agent(s) in question (windows control panel, ip phone settings, headset settings, volume controls in soft phone ... everything) and any configuration files you may have modified from "stock".

That being said, there are some interesting tools available since asterisk 1.2 in meetme, that may shed some light on your issue ... experiment with this (as both agent and prospect) and see if it is possible to make "adjustments" to your sound ...: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe

fibres wrote:Hi Guys

Having a strange issue at moment.

When we dial out from vicidial the customer is really quiet and the agent struggles to hear them.

When monitoring using the blind extensions monitoring and listening to call recordings the agent is clear as crystal but the customer is really quiet.

If we dial directly from a softphone, therefore bypassing the vicidial and meetme conferences the customer is fine.

Is there any way of increasing the gain on the customers call from within the conference?

Regards

what codec are you using from dialer to termination and from the softphone to the dialer?
whats the load average of the server?
does your headset have noise cancellation?
is your setup behind NAT?

Hi both

Well boybawang

We are using G711alaw all way through at moment

The highest load avg I seen on server is about 2.8 and this is running on a dual quad core, that for some reason suse linux seems to see as 16 cores.

No noise cancellation on the server

Yes the setup is behind NAT.

Hi William

The installation is stock. Done no messing. Well I did import a database from a previous installation having done an upgrade on the script.

Yes I have been using vicidial for a while and picked up a lot. But wouldnt say im dangerous!

I would look at the settings on the agents pcs as suggested, however calls which are made direct from the softphone as in not rhough vicidial go through fine.

This seems to be an issues related to meetme.

Regards

We are also experienceing some very strange issues with delay and echo.

Dialing out through eyebeam direct works great, through vicidial and we get echo and delay.

The really strange one is if we log into the call through blind monitoring it sounds fine.

If I log in as an agent and make some calls with 2 headphones on, into two computers one logged in normally and the other as a blind monitor using the 08600xxx extension then on the monitor headset I hear the customers reply instantly but it has a 2-4 second delay to come through to the agent headset.

How does blind monitoring work? I know it has something to do with a IAXloop and packet grabbing, does this mean it is bypassing the meetme conference?

Regards

Below is the output from a prolonged run of zttest with agents on the system


--- Results after 4561 passes ---
Best: 100.000 -- Worst: 99.602 -- Average: 99.968471, Difference: 99.998689

Can anyone tell me if this is acceptable or could this be an issue?

Regards

describe your network ... everyone in the same room, on the same network, as the server?

check jitterbuffer (which can cause delays while reassembling the sound packets to maintain the correct order of transition)

Hi william

Our server is hosted in a datacentre on a 100mb connection. This is a high class hosting with SLA and bandwidth guarantess.

Our site has a 100mb Leased line which is tested as capable of pulling its full capacity. We are currently using around 4-8mb a sec on this

all agents are in same room attached to this line via a cisco router running NAT.

Calls direct not through the vicidial are fine. Calling through vicidial has the issue.

Regards

describe the hardware it's installed on, and the server load when this is happening.

how many agents do you have at a time? call ratio? specs of server?

The server load never goes above 3.0
We are running on a dual quad core xeon 2.5ghz
8gb ram

We have about 15 agents on with a dial ratio of 4

We are not doing call recording

We seem to have resolved the echo and delay by switching from eyeBeam to Zoiper so suspect it was some kind of memory leak in eyebeam as we discovered that would be fine for first 5 mins after reboot but then it would start to have a delay which would increase.

We are still experiencing low volume of customer.

Regards

Vicibox 3.1 ISO install.
Vicidial 2.4 SVN

Consider using zypper to update to the latest (just in case) we are on 3.1.9 now.

Please ALSO post your BUILD from Vicidial (not likely an issue, but this is the developer trunk and you'd be amazed how often listing that can get a response like "we just fixed that, update" if/when it's appropriate, from Matt ). So help make that possible, along with tracking bugs with builds, by always listing your vicidial version WITH BUILD.

Have you considered (just to be funny) changing your agents to IAX to see if that makes a difference (since changing your phones already made one difference ...). No, you do not need to change anything in your trunks when you change your agent protocol.

Hi William

We are actually using 3.1.8 I think. Downloaded latest iso 2 weeks ago.

I am considering re-installing the systems to use the 64 bit version this weekend so will use 3.1.9

How do I check the version of trunk im using?

I will try one or two agents on iax see if this makes a difference.

Regards

bottom left corner of every admin page. even if you skip other stuff, that is important for bug-tracking (and often will pop an answer immediately if it IS or WAS a known bug in your version, or has recently crept up and is getting attention)

VERSION: 2.4-301
BUILD: 110215-2135

Should be in my signature now. Just i case it hasnt update yet here it is.

Regards

I think you should add a USB timer or TDM400P to increase timing accuracy. The zttest result should be above 99.98%.

as a rule, sound at the recipient side is managed by the sound card at the recipient. the channel is not altered in the asterisk system unless it is a dahdi channel. do all agents experience this? are they all on the same hardware? headset?