Voice of the Recipient is very Low, crack and choppy

I am having a problem in my server.

when i am calling outside the voice is very low and can even hear the person which i am calling.

how to increase the volume aside from the phone itself is there any other way to increase the volume?

You have not told us anything about our system so we have no way to help you. I am going to guess that you are using analog trunking so you might investigate how to increase the volume on the analog adapter card.

Yeah exactly sir. i am using sip trunk. VoIP Linksys Sipura SPA-3102.
how can i increase the volume.

all i can see in the internet is,increase the tr and rx gain.
but i didn’t get the correct parameters.

Did you try any other phones? I don’t, and I am sure others don’t want to ask multiple questions. If you want some help, you need to tell us your setup, details on the network, your FreePBX version and where you got it. The more details you give, the more people will chime in.

Also, a quick search on Google for this phone and issue, pulls a lot of complaints on low audio volumes.

the tx/rx gains impedance sample size and other parameters of the 3102 can be set either from the built in http server and/or the config file provided by a tftp service. here are no settings for crack or choppiness, that is a network issue.

Bing can only come up with xhtml or asp.net at first glance.
if thats the case you are in the wrong forum

“the tx/rx gains impedance sample size and other parameters of the 3102 can be set either from the built in http server and/or the config file provided by a tftp service. here are no settings for crack or choppiness, that is a network issue.”


Given your recent:-

some of us are probably a little confused with some of your posts, please elucidate in what you believe is the “right forum” . . . .

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Sorry for the late reply guys.
so this type of Voip is having this problem?
My Elastix Version is 2.3 and asterisk is 1.8.

is there anyone from here in the forum guys experiencing this kind of a problem,
what is the best solution to fix this one,

in my network i am having a 16mbps connection so it is not the network issue i think.

thanks for your responce guys.

we can’t support elastix here because they do it their own way and your versions of freepbx and asterisk are hardley supported they are years old. but if your calls are going through the 3102 there is nothing that freepbx nor asterisk can do to change its misconfiguration.e fix that first.a

thanks for the response dicko.

i dont know if the issue is in 3102 or in the telephone,
my ip telephone is zed 3 cn2x4.

i usinf also freepbx also,because in elastix there is Unembedded freePBX.

i log in there also.

If your call is using a “landline” then you are using the 3102, if not then try another phone, you will find that asterisk itself and probably all SIP providers never need tx/rx re-set from 0db .

but if you think otherwise, and your version of FreePBX supports SIP settings then you could play with adding there:-


I believe the valid range is -13(db) to (+)13(db)


hi dicko can please tell me where i put this code?

is this the exact parameters.

thanks for the help dicko:)

is the call over a landline?

I can solve this problem for you in 3 easy steps.

  1. Get rid of the SPA3102 and analog line.
  2. Port your number to a SIP Trunk Provider
  3. Never use SPA3102 and analog line configuration again.

Thank me later.

For SIP Trunk providers: checkout Schmooze SIPStation for easy setup, I use PhonePower because they are very cheap ~$8/month if you prepay 1yr, but the setup to FreePBX is technical and not supported

Dozens of hours of frustration with SPA3102, SPA8800s and analog lines with VoIP.

Seriously though, its not even worth troubleshooting. Get a SIP Trunk. It’s cheap and painless.

yeah, through the provider.sip trunk.
so which is the better voip should be used?
aside from SIPSTATION?

You’re dealing with two problems. The first is the volume, which could easily be a function of the phones (the SPAs) or the analog card TX and RX settings. The other problem you are contending with is choppiness. The first is usually solved through step-wise refinement of the setup, adjusting the send and receive gain to the point that the system will actually work the way you want it.

The second (choppiness) can come from the lines (if you are on a slow or laggy network connection) or through your hardware (not enough horsepower to get the packets delivered). If you’re connected through DSL (for example) and are on a 1.2M down/384K up circuit and are transmitting a lot of stuff, you could be running out of network. If you’re doing something like recording a lot of calls, you could be running out of disk or interrupt power. It’s really hard to troubleshoot these things, though, since our only interface to your system is your posts.

SIPStation is a sponsor of FreePBX, so they’re always a good resource. I’ve used a few other SIP providers, and if you’re looking for comparables, I like Voip Innovations. Of course, your cost is going to depend on what services you ask for - the more you get, the more it ends up costing. I have about 20 numbers, some with 800 service, plus FAX, T.38, SMS, and a bunch of other stuff and I think I’m getting a really good deal.

I’ve never been a fan of the SPA phones - I use Cisco 7940 and 7960 series phones and install the SCCP-Chan-B software. I even wrote a quick description on how to use and configure the software with FreePBX. These phones are good enough for the CTU - they’re good enough for me.

As far as analog lines go, I’ve used lots of analog interfaces and never really had any trouble with them (discounting the problems with the Rhino cards I bought - Hi James!). I’ve got one system that connects 16 analog phones to a server, recording every call, and sending all of that out over a SIP connection. We get choppiness once in a while, but it’s usually associated with one of the managers watching Hulu on his desktop.

Keep at it - you’ll get it.

hi cynjut,

it was nice sharing what you have in your system,
before its working great,then this few weeks it happens,the voice is low,choppy voice,crack voice.
and its hard also for me because i am not the one who configure install my freepbx,so i doing self study on it.
it was nice to hear there system configuration here in the forum.
admins are helpful, and even users here.

thanks for sharing your knowledge,system here.
now i will do this tx and rx volume,
i will create a new parameters on it.
i will try if it will works.

i found the solution i increase the number in web gui of the SPA 3102.

this is the forum that i found and increase the voice.

I have found much information about how to make calls from Asterisk over the PSTN line and SPA3102

Are you using Asterisk or a GUI flavour of asterisk?

Either way try this guide as it will give you enough info to work with – it works well and I have set it up in the past for clients who did not want to pay for a telephony card and only wanted a fail over line.


After that guide Australian settings as below.

Under PSTN line at the bottom.
FXO port impedance – 220+820||120nf
SPA to PSTN Gain – 3
PSTN to SPA Gain – 3
On Hook Speed – 26mv Australia

Under Regional
FXS Port Impedance – 220+820||115nf
FXS Port input gain – 0
FXS Port Output Gain – 0
More echo suppresion – no

If you have problems with echo and low volume try adjusting the gain settings under the pstn tab. Success will depend upon your line quality more than anything and it can be a very fine balancing act.

Once you have sorted out the other issues a google search for Australian call progress tones will tell you what to

If you can afford it buy a hardware card such as sangoma or digium (openvox is an option too at the cheaper end of the market) as these are proven peformers and always work if setup correctly.

Glad you followed my advice, Personally I would say that the 3102(and 200x without FXO ) are also proven performers , I have many I use for connection to physical fax machines, they are not dependent on hardware cards, they are highly configurable and as you have found out after setting them up correctly thet wotk correctly , they do t38 and you can buy the off market ones (thanks Cisco for buying linksys, AND thanks linksys for buying Sipura;-) ) on ebay for a few bucks. As a side note you can significantly improve the performance on the SIP side by setting the SDP packet size to 20ms to suit asterisk.