Hi,
I am new to Asterisk, I install it successfully but I have crack sound and lot of noise during conversation.
My IRQ are note shared. I have only one X100P card (I must say it is a clone because I wanted to try before setting a serious environment in place)
I do not run any X-Windows on the box.
Any Idea ? Is it normal ?
It could be normal.
How about giving us some data?
What phone are you using?
You told us how you are connected to the telephone network, but what kind of phone are you using to call out?
I hope it is not normal … or Asterisk is not really usable because crack noise is so loud …
Here are some more informations :
I am running latest CVS version of Asterisk :
Asterisk CVS-HEAD-04/10/05-08:12:28 b
The hardware is a Clone bought from ebay X100P FXO
My box is an Athlon 2800 with following devices:
0: 33205356 XT-PIC timer
1: 8 XT-PIC i8042
2: 0 XT-PIC cascade
3: 6132162 XT-PIC uhci_hcd, uhci_hcd
7: 33154301 XT-PIC wcfxo
8: 1 XT-PIC rtc
9: 0 XT-PIC acpi
10: 792208 XT-PIC VIA8233
11: 21564 XT-PIC eth0
12: 92 XT-PIC i8042
14: 110334 XT-PIC ide0
15: 596000 XT-PIC ide1
I am running X-Lite Soft client on a computer on the
same local network. No firewall between the two client and Asterisk box.
Is it related to the hardware of my configuration ?
It is probably the x100p clone. YMMV with these cards. For testing purposes they work well but if you want to use it in a live/production environment where the quality matters then you should pony up for a better FXO/FXS.
Which hardware do you suggest me ?
Also I would like to be sure it is an hardware problem before buying a better card …
This is the best entry level FXO/FXS card
digium.com/index.php?menu=devkit-fxofxs
You could also try this, but they are pretty new:
store.yahoo.com/grandstream-netw … nphad.html
You can use some softphones (X-Lite) to see if it is problem the server or Asterisk config.
Thanks for a bit more info.
It would have been good to know which distribution / kernel you use.
Since you are using a soft phone let me speculate:
If your kernel is 2.6 you are probably using ALSA.
If I use a softphone (any) under linux I get crackling noise that makes it unusable. If I load the OSS emulation modules and configure the softphone to use OSS the sound is good.
Softphones will not make you happy. I have not seen one that works well for me.
About your FXO card: you get what you paid for. I started with one, and it kind of worked. Not enough volume or too much echo.
I am currently using a grandstream ht-488 and am fairly happy with it. Excellent sound quality.
In short: if you are using 2.6.x with ALSA and a softphone then the crackling is ‘normal’. Is it * fault? No.
Hello,
Thanks for your reply.
That’s true cheap card might be less reliable but crakling noise might come from som many reasons that I want to be sure before buying other hardware. Nevertheless, I am already convinced about the value of Asterisk software at beleive I will extend my project further !
To respond to your question :
Asterisk is running on a Linux platform :
Linux localhost.localdomain 2.6.10-1.741_FC3 #1 Thu Jan 13 16:38:22 EST 2005 i686 athlon i386 GNU/Linux
X-Lite Softphone is running on a Windows XP Home
Is the client or the server responsible of this crackling noise. I beleive the server is because other clients with VOIP thechnologies (skype) do not have anny issues on my windows platform. I desactivated speech recognition which is a big break for this kind of sofware !
From what I see ATA-488 is a standalone device ?!
There is no point in evaluating Asterisk sound quality using a soft phone. Steal, borrow, or buy a SIP phone.
Or use an ATA with an analog phone.
As to the ht-488:
It is a little box with two phone ports. They are sip clients.
You create two 'friend’s accounts in sip.conf.
It hooks up to your ethernet.
On power fail the FXO port is connected directly to the FXS port.
When powerd, you can have the FXO ring through to the FXS without Asterisk even knowing there is a call, or you can have the FXO call an extension and then have asterisk do whatever you need. The FXS dials Asterisk (if that is what you configured) and you can connect the call via * to your sip provider, or out the FXO port. You can also dial *00 and connect the FXS to the FXO for pots dial tone.
I am working on a review of the unit. There are some issues with stability of the firmware. This is a new toy. Let me get a feel for it. If it really works well, it would be hard to beat!