vicidial.org

Dear All, I come from China, I'm here to asking for your help. I build 1 server for a call center to use. But there one problem can't not solve, the sound quality is too bad.

we are using a dual core e4600, 4g mem, solid hdd, T1 card. G729 code.
25 uses.

i modify the rxgain & txgain, it seems have no any help to improve the sound quality. Do you know where I can modify to improve it. Thank you!!

why are you using g729 with a T1? what model T1 are you using and who is your provider?
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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.

Similar to This:

GoAutoDial ?.? CE | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

hi, thank you for your reply. Actully, it have a litte complicated. I have 2 server. Server A place in Hong Kong, with a Sangom A200 2 port of T1. Then I have another server B place in china mother land, they connected with a 4M IPVPN(MPLS). 2 server connect with IAX2. the user is in China using server B. and is using g729. All user loggin server B to dial through server A. I'm using g729 just because I test like ulaw, and other, g729 is better.

I just a new fish of asterisk, all the user report the sound quality is too bad, some times can hear the echo, and not clear. And in the same time, we have another PBX system, it works fine. I 'm wondering whether asterisk is not setting right or not.

vicidial is not designed to run with geographically separated servers in the same cluster. if indeed you have a cluster.

which brings me back to:

Please describe your installation of Vicidial. While it's nice that you mentioned your processor model number and how much memory you have, you have not given the basic information regarding Vicidial which is required for assistance.

Now that you bring up that you have a distributed system, describing the network topology and which processes are being used in which servers would be helpful.

To discuss sound quality, we need to know the exact path the calls are taking.

prospect -> (PSTN) voip provider (SIP) -> (SIP) server XX via SIP -> server YY via ...??? -> user ???

Thank you very much, William.

Server A in HK:
cpu: Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz
ram: 4g
with viciaialnow ce 1.3

server B in china:

cpu: core duo 2 T4600 @ 2.2Ghz
ram: 4g
OS: goautodial ce 2.0


pstn->serverA-----iax2----->serverB-----sip----->X-lite

Maybe helpful to you.

where do you experience the call quality issues? can the agents hear sounds generated by each of the vicidial systems clearly? (this would be something you could test)

how good is your bandwidth at each leg? dropping any packets? lag time? jitter?

the voice is not big enough, and the voice echo, mix with noise. The client's PC is AMD 240, 2g ram, with a professional headset, x-lite. XP os.

below is the ping capture





[root@go ~]# ping 192.168.1.40
PING 192.168.1.40 (192.168.1.40) 56(84) bytes of data.
64 bytes from 192.168.1.40: icmp_seq=1 ttl=63 time=14.6 ms
64 bytes from 192.168.1.40: icmp_seq=2 ttl=63 time=14.7 ms
64 bytes from 192.168.1.40: icmp_seq=3 ttl=63 time=14.6 ms
64 bytes from 192.168.1.40: icmp_seq=4 ttl=63 time=14.6 ms
64 bytes from 192.168.1.40: icmp_seq=5 ttl=63 time=14.6 ms
64 bytes from 192.168.1.40: icmp_seq=6 ttl=63 time=14.8 ms
64 bytes from 192.168.1.40: icmp_seq=7 ttl=63 time=14.7 ms
64 bytes from 192.168.1.40: icmp_seq=8 ttl=63 time=14.7 ms
64 bytes from 192.168.1.40: icmp_seq=9 ttl=63 time=14.6 ms
64 bytes from 192.168.1.40: icmp_seq=10 ttl=63 time=14.6 ms
64 bytes from 192.168.1.40: icmp_seq=11 ttl=63 time=14.6 ms

--- 192.168.1.40 ping statistics ---
11 packets transmitted, 11 received, 0% packet loss, time 10003ms
rtt min/avg/max/mdev = 14.603/14.683/14.836/0.066 ms

Can you check the load average on your GoAutoDial server when your agent experiences voice quality issues? Can you also post the output of this command:

dahdi_test -v

williamconley wrote:where do you experience the call quality issues? can the agents hear sounds generated by each of the vicidial systems clearly? (this would be something you could test)

how good is your bandwidth at each leg? dropping any packets? lag time? jitter?

1) you did not post testing sound to/from each of the servers, you still only mentioned total throughput. troubleshooting often involves extra work to break down a problem to components. this can be tedious (but noone ever said troubleshooting was a party)

2) check your bandwidth using a proper tool such as wireshark. pinging is not ... a way to measure sip or iax packet throughput or jitter or latency.

gardo wrote:Can you check the load average on your GoAutoDial server when your agent experiences voice quality issues? Can you also post the output of this command:

dahdi_test -v

Hi Thank you,



[root@vici ~]# zttest -v
Opened pseudo zap interface, measuring accuracy...

8192 samples in 8192 sample intervals 100.000000%
8192 samples in 8190 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8192 sample intervals 100.000000%
8192 samples in 8192 sample intervals 100.000000%
8192 samples in 8190 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8194 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8190 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8192 sample intervals 100.000000%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8190 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
--- Results after 20 passes ---
Best: 100.000000 -- Worst: 99.975586 -- Average: 99.987183

williamconley wrote:

williamconley wrote:where do you experience the call quality issues? can the agents hear sounds generated by each of the vicidial systems clearly? (this would be something you could test)

how good is your bandwidth at each leg? dropping any packets? lag time? jitter?

1) you did not post testing sound to/from each of the servers, you still only mentioned total throughput. troubleshooting often involves extra work to break down a problem to components. this can be tedious (but noone ever said troubleshooting was a party)

2) check your bandwidth using a proper tool such as wireshark. pinging is not ... a way to measure sip or iax packet throughput or jitter or latency.

Hi, Thank you very much. in the beginning I'm wondering whether the netwrok not fast enough to cause this problem. So I buy a new VPN line (MPLS), it is the best vpn. but the problem still exist. In the same network, I'm using a TOSHIBA PBX system, but it is ok. and I don't know where I can modify the system to make it run better.

GoAutoDial doesn't use Zaptel (zttest). It uses DAHDI by default (dahdi_test).

LazyPaul wrote:

gardo wrote:Can you check the load average on your GoAutoDial server when your agent experiences voice quality issues? Can you also post the output of this command:

dahdi_test -v

Hi Thank you,



[root@vici ~]# zttest -v
Opened pseudo zap interface, measuring accuracy...

8192 samples in 8192 sample intervals 100.000000%
8192 samples in 8190 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8192 sample intervals 100.000000%
8192 samples in 8192 sample intervals 100.000000%
8192 samples in 8190 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8194 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8190 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8192 sample intervals 100.000000%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8190 sample intervals 99.975586%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
--- Results after 20 passes ---
Best: 100.000000 -- Worst: 99.975586 -- Average: 99.987183

At the risk of repeating myself while repeating myself:

LazyPaul wrote:

williamconley wrote:

williamconley wrote:where do you experience the call quality issues? can the agents hear sounds generated by each of the vicidial systems clearly? (this would be something you could test)

how good is your bandwidth at each leg? dropping any packets? lag time? jitter?

1) you did not post testing sound to/from each of the servers, you still only mentioned total throughput. troubleshooting often involves extra work to break down a problem to components. this can be tedious (but noone ever said troubleshooting was a party)

2) check your bandwidth using a proper tool such as wireshark. pinging is not ... a way to measure sip or iax packet throughput or jitter or latency.

Hi, Thank you very much. in the beginning I'm wondering whether the netwrok not fast enough to cause this problem. So I buy a new VPN line (MPLS), it is the best vpn. but the problem still exist. In the same network, I'm using a TOSHIBA PBX system, but it is ok. and I don't know where I can modify the system to make it run better.

You can keep shooting from the hip, of course, but since you have no target to aim at ... your odds aren't good. I do wish you the best of luck, though.

Thank you william, you really a big help. it seems no way I can do. Do you know where the goautodial system can modify something like: rxgain & txgain ? this just make the voice a litter bigger, but can't make it better.