Problem in voice

[AGA]

Hi,
i have an issue with a new installation of a clustered Vicibox VERSION: 2.4-350a , for a new site for 150 agents
- Dialer bi processor Xeon Dell T610, 32 G Ram, HD 300 15 KRPM
- Web, Dell PE110, Xeon, 8 G Ram
- 3 dialers, Dell PE110, Xeon, 8 G Ram
- 1 Storage server Dell PE110, 8 G Ram, all calls being recorded
- 2 Freepbx servers with digium cards with IAX trunks on each dialer
I followed step by step the Poundteam manual and everything went smoothly. Many thanks for this great doc ! I can log my zoiper on each dialer and make manual call but when I use ratio dialing mode (ratio 2) the voice turns terrible. We can’t understand the customers and they can’t understand us. The system get very slow
I checked that codecs were the same on freepbx servers and dialers but no luck.
Would anybody have an idea ?
Many thanks in advance

Antoine

[williamconley]

1) Welcome to the party

2) Your version information is helpful, but please provide it in the following manner: 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 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

(Primarily: the build is missing from your vicidial version, you did not mention the vicibox installer version, two of the most important pieces of information to track bugs through this free forum ...)

3) you mentioned something about freepbx and digium cards ... but if you are using iax there is no need for digium cards ... and you did not mention the type or model numbers of the cards.

4) If you used vicibox to install, you should not have needed the manual for clustering as Vicibox since 3.0 has clustered during installation automagically (thanks to Kumba), so we have some concerns to address there ...

5) Please list the path of these calls from the agent to the client (and the meetme room in there somewhere ...). Are the trunk dialers also the agent dialers (ie: agent meetme room is on the same dialer as the outbound phone call?)

6) you gave specs for 4 dialers (3 +1) plus a web, a storage and two freepbx servers ... I presume there's a database server in there somewhere? LOL

7) server load on each of the servers during the issue? (especially the dialer upon which the trunk and agent's meetme room reside)

8 ) Bandwidth available that is EXCLUSIVE to VOIP (ie: NOT in use by any other processes, users, agents, servers)? total number of calls in progress when the problem occurs?

[AGA]

First of All,, happy new year to all vicidial users! hope 2012 will bring a lot of good calls to all companies using this great software !
Thanks William for taking time to answer me.
2) below the information requested :
I installed all the servers with ViciBox_Redux.i686-3.1.14.iso and used the latest SVN available. I ended with VERSION: 2.4-350a BUILD: 111201-0939. Asterisk 1.4. No extra sofware installe after installation.
I installed full version on all servers then followed the PoundTeam tutorial
3) My hardware is like this
- DB bi processor Xeon Dell T610, 2*Intel® Xeon® E5607 Processor 32 G Ram, HD 300 15 KRPM (this is the one !)
- Web, Dell PE110, Xeon, 8 G Ram
- 3 dialers, Dell PE110, Intel Xeon E1220 ((3.1GHz, 4C/4T, 8M
Cache, 80W, Turbo), 8GB RAM ,
- 1 Storage server Dell PE110, Intel Xeon E1220 ((3.1GHz, 4C/4T, 8M
Cache, 80W, Turbo), 8GB,, all calls being recorded
- 2 Freepbx servers with TE420B digium cards
I have 8 E1 in my installation. They are connection in 2 TE420B Digium card with Echo cancellation. Each card is installed in a Freepbx server. Each dialer is linked with FreePBX using an IAX Trunk;
All servers are in the same IP range and Same Submask.
Digium cards are working ok, i can connect a softphone on FreePBX and make calls normally. I can also connect a softphone on a ViciDialer and make manual calls without any problem. The problem occurs when i log in the agent interface and use ratio mode. I made test with 2 agents, with a ratio of 5, full recorded calls. The load on the dialer used was 0.1 and 0.12 on the Freepbx
5) the meetme rooms are created on the same than the dialer
the brandwith is the one of my LAN 100 mbs as i do not use VOIP to make calls but E1.
Many thanks in advance !

Happy 2012 !!
Antoine

[williamconley]

thanks for including your "build" this time with your vicidial version, please include your FULL asterisk version (since you installed with vicibox, it's likely that one ... but for non-vicidial surfers who land here, and thanks to google this WILL happen, please include the full version of asterisk so MORE will land here with that version of asterisk ... always list full versions in a technical support forum as that is a technical detail that is useful)

on that note: moving this topic to the support board where it belongs. LOL

if you are experiencing problems with call quality using digium cards and linking the servers with iax try:
1) link between servers with sip instead of iax for testing (just one agent to see if that one is any different)
2) try a dial ratio of 1 to simulate manual dialing
3) do your digium cards have echo cancellation?
4) put the E1 card in a vicidial box and see if that removes quality issues
5) put the E1 card in a NONFREEPBX machine (pure asterisk, no overhead) and see if that removes the quality issues
6) as freepbx has A FREAKIN LOT of overhead, try skipping the freepbx dialplan and just using it as if it were a pure asterisk machine. allow the call to pass through without using the freepbx dialplan, ONLY use the freepbx created trunk to the e1 card (single line of execution: dial!)