[Asterisk-Users] Static on inside end of conversation

[Jeff Busch]

Update on this... And it is still not solved.

This is actually fairly interesting.  I have two installations at a
construction company.  They are both running similar class machines (I
was wrong in my initial post) they are:

System "A"
2.4 ghz Celeron
1 gb RAM
IDE Drives
Asterisk at Home 1.13 (Asterisk 1.0.9)
An Audiocodes MP-108 to interface with PSTN
9 Polycom IP-500
Phone system is on 16 port linksys switch.  All PC's are on separate 16
port switch.  Both switches are plugged into our VPN firewall (Netscreen
5xp)


System "B"
2.4 ghz Celeron
1 gb RAM
IDE Drives
Asterisk at Home 2.1 (asterisk 1.2)
5 Sipura SPA-3000  to interface with PSTN
12 Polycom IP-500
Phone and data on NEW Netgear 48 port Smart Switch.  Voice is on Vlan 02
with QoS set to HIGH.  Data is on Vlan 01 with QoS set to LOW.  Switch
is connected to Netscreen 5xp VPN firewall


Polycoms are running the newest possible firmware for this phone:
Bootrom 2.6.1 and SIP 1.5.2


Everything is working fine on both systems, but BOTH systems are
experiencing echo on the IP side of the conversation.  Some phones are
worse than others.  

System A is experiencing echo both on internal Station to station calls,
and on outbound calls.
System B is experiencing echo on internal station to station calls, and
no one has complained about echo on outbound calls yet.

The Polycoms are running G.711u as their 1st configured codec (G711a as
#2 and G.729AB as #3) and Asterisk is configured to disallow=all &
allow=ulaw.

I have searched and searched concerning echo and Polycom phones and
haven't found anything that seems to be relevant.  Any help would be
appreciated.

One more note, I am NOT running TDM4xxp cards on either of these
machines, so Zaptel information concerning echo wouldn't be relevant,
correct?

Thanks!


Jeff Busch 


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike
McMullen
Sent: Tuesday, November 29, 2005 6:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on inside end of conversation


----- Original Message -----
From: "Jeff Busch" <Jeff.Busch at lewisbuilds.com>

> Hello,
> 
> I am running the following configuration:
> 
> 2.8ghz P4 with 1GB of RAM
> Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones 
> Asterisk at Home 1.3 (this is Asterisk 1.0.9)
> 
> End users are complaining of an echo and static on the inside end (the

> internal side), but the outside end of the conversation doeesn't 
> notice anything.
> 
> Does anyone have any suggestions on troubleshooting / fixing this 
> problem?
> 

Hi Jeff,

I recommend upgrading to Asterisk at home 2.0 which was just released. 
It uses Asterisk 1.2 and a 2.6.9 based kernel which handles i/o and
interrupts much better.

While the link below discusses issues with digium cards, in general the
interrupts and IDE vs SATA drive discussions are of use no matter what.

http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te40
5p_noise.html

I have 2 Digium T400P cards connected to 8 POTS lines and 10 sipura
spa-841 phones.

I went through two PCs we had (2.9 GHZ celeron, 2.1HGZ Athlon
XP) both with IDE drives. I finally declared war on echo and the Rice
Krispies syndrome (Snap, Crackle, Pop) on the internal end of the
conversations. 

I went and bought an ASUS P5LD2 motherboard with 1GB memory, 3.2GHz  P4
with hyperthreading and a 2MB cache, and a SATA drive. I installed
Asterisk at Home 2.0. After some minor magic to get the correct merlin
gigabit ethernet driver for CentOS 4.2, everything came up perfect.

This was my thanksgiving project and so far sound quality has been
perfect. No echo and no rice krispies.

I simulated network load on the system by copying multi-GBs of files
through the net from another server with scp while I called out and back
into the system on multiple lines. Even with the scp reporting
9.5MB/sec, the phone sound quality was fantastic.

I then upped the ante by copying multi-gb files on the hard drive which
when viewing stats with top (hyperthreading shows as 2 CPUs and you run
the SMP kernel) both CPUs showed no idle time. IO- Wait state never
greater than 10%. Phone calls were still perfect with no echo or noise.
Out of the 10 vmails I left as part of the test, only one had
3 very faint pops in a 30 second message. They could have come from the
POTS line for all I know.

I ran extended phone conversations by calling the Asterisk system from
our old phone system, picking up the extension on the called SIP phone
and then playing Law & Order dvd episodes (lots of talking) and placing
the handset near to the speaker and taking the old phone system handset
and listening and talking back into it for 30 minutes at a time.
Necessity is indeed a mother.... ;-) 

The calls were perfect. I'm amazed at how clear the dvd audio came
through. 

I then reversed the process and played the dvd audio through the
Asterisk system handsets while listening and talking back through the
old phone handsets.

After 30 minutes the quality was still excellent.


Hope you find some of this ramble useful.

Mike
 

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users