Bandwidth limited?

[legin]

Hi everyone, I have an GoAutoDial server which has about 20 agents logged in... I have also a 4mb Bandwidth, but only using half of the bandwidth, my question is: There is a way that I can make the server use all my bandwidth? because I'm having troubles with the calls, the clients don't listen well to my agents, and I thought it could be the bandwidth... I'm using the g729 codec, but when I put "g729 show licenses" in CLI, it says "No such command 'g729 show licenses' (type 'help g729 show licenses' for other possible commands)"
Can you help me please, I really want my system to work well....

Thanx!!!!

[williamconley]

1) Welcome to the party!

2) when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) How do you know you are using g729?

4) If you are using unlicensed g729, that command will never work anyway.

5) "The clients don't listen well to my agents" is not extremely descriptive. Sounds like they just don't follow instructions unless that's a language barrier issue in your posting. Are you saying that the volume is too low? Sounds is garbled? Robo-voice? Missing pieces?

6) Your Carrier settings "allow/disallow" will determine your codec. allow=ulaw instead of allow=g729 would change your codec from one to the other ... but I'm not certain you are even using g729 to begin with. Post a bit more detail ... (making sure you're not an amateur, one way to tell is the ability to post details, and if you are an amateur you could be using gsm and not know it! LOL)

[legin]

Hi william.... yes, I have a language barrier, I am a spanish spoken person, so....

I'll go steps by steps ok?

1) Thank you!

2) GoAutoDial CE 2.1 | Asterisk 1.4.39.1-vici | Single Server | Twinkle Soft Phone | No Extra Software After Installation | Intel(R) Pentium(R) 4 | CPU 3.00GHz

3) I have configured in my Trunk like: disallow=all allow=g729
So I guess I'm forcing it to use the g729 right?

4) Is not an unlicensed G729, coz I bought several licenses

5) With "The clients don't listen well to my agents" I mean choppy calls

6) In point 3 I guess I just ansered this question no?

Thank you very much for your help and sorry for the first post, as you say I am very amateur with this!!

Again thanx!!!!

[gers55]

buenos tardis

I don't think bandwith is your issue.

unless your server is hosted i would turn agents back on to ulaw,alaw codec as william says with allow=ulaw,alaw .G729 uses cpu which will affect the load on your server.

I would also check the call quality with 10 agets logged in and check your server load. I think you are pushing things with 20 agents on a single pentium 4 processor

[williamconley]

Of course, before changing anything you should confirm what you are actually using. Setting allow= usually works fine, but CHECKING is best to be sure.

Code: Select all

asterisk -rx "sip show channels"

This will actually tell you to the codec in use, regardless of what you "requested". Often there are errors ... checking can help find them.

Test with SIP and IAX2 (both protocols) each with codecs: ulaw, gsm, and g729. If any of them are good, of course, just go with it. LOL

Remember to check these changes on both the carrier leg of the call (prospect) and the agent leg of the call (but not g729 if you are using a free agent soft phone, because free ones don't have g729).

Also remember that both protocol and codec are independent between carrier and agent (a SIP carrier on g729 can talk to an agent using IAX2 and ulaw, they do not need to change together, asterisk mixes them inherently).

Best quality should be attained with ulaw on all legs unless you have limited bandwidth. Easy to test with a single call, then ramp up and see if things fall apart under heavier load. This can indicate a load issue in the server, a hardware issue in networking (ie: a router that can't handle the traffic) or a bandwidth limitation. It has even been known to reveal a bad network component at the ISP level (I've had a technician from Brighthouse fix a router connection down the block to resolve an issue in a client office!).