[zz000mm]

Hi all,
We are using Asterisk@home with the Freepbx front end to run our small office telcoms. We are using an AMD Attlon XP 1800+ 1.52 Ghz. We have to Sip accounts that are monitored and four extensions, two internal via a Sipura SPA 2100 and two remote via Com-on-air voip/dect controllers.

Everything has been reasonably working fine although it has been a real headache over the last year learning about asterisk/phone adapters/routers/Nats etc

Anyway, we find that the calls are initiated or picked up fine with good audio quality but after a minute or so the voice quality starts to fall apart, coming and going with the other party saying it sounds as if we are robots....

I've attached a copy of the Asterisk Sysinfo and Asterisk info below. Our physical memory is running at 97%. I do not know what the implications of this are.

Our sip.conf looks like

register => **************
register => **************
externip=***************
localnet=192.168.0.0/255.255.255.0
nat=yes
canreinvite=no

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
qualify=no ;new
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm



Any ideas why our sound quality is having problems? We have not registered for the Digium codec. Should we?

Any help appreciated.

Asterisk Sysinfo is below
**************************************************



ERRORS
File Line Command Message
common_functions.php 294 file_exists(/proc/scsi/scsi) the file does not exist on your machine



System Information: localhost (192.168.0.6)
System Vital
Canonical Hostname localhost
Listening IP 192.168.0.6
Kernel Version 2.6.9-34.EL
Distro Name CentOS release 4.3 (Final)
Uptime 3 days 15 hours 36 minutes
Current Users 1
Load Averages 0.00 0.00 0.00



Network Usage
Device Received Sent Err/Drop
lo 9.38 MB 9.38 MB 0/0
eth0 313.73 MB 298.86 MB 0/0
sit0 0.00 KB 0.00 KB 0/0


Hardware Information
Processors 1
Model AMD Athlon(tm) XP 1800+
CPU Speed 1.52 GHz
Cache Size 256 KB
System Bogomips 3033.76
PCI Devices - Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+
- IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE
- Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller
- VGA compatible controller: VIA Technologies, Inc. VT8378 [S3 UniChrome] Integrated Video

IDE Devices - hda: WDC WD400EB-00CPF0 (Capacity: 37.27 GB)
- hdd: E-IDE CD-ROM CR-836S

USB Devices - (3x) Linux 2.6.9-34.EL uhci_hcd UHCI Host Controller
- Linux 2.6.9-34.EL ehci_hcd EHCI Host Controller




Memory Usage
Type Percent Capacity Free Used Size
Physical Memory 97% 5.71 MB 212.72 MB 218.43 MB
- Kernel + applications 41% 90.05 MB
- Buffers 19% 41.58 MB
- Cached 37% 81.09 MB
Disk Swap 0% 760.88 MB 0.00 KB 760.88 MB



Mounted Filesystems
Mount Type Partition Percent Capacity Free Used Size
/ ext3 /dev/hda2 4% 32.76 GB 1.27 GB 35.85 GB
/dev/shm proc none 0% 109.21 MB 0.00 KB 109.21 MB
/dev/shm sysfs none 0% 109.21 MB 0.00 KB 109.21 MB
/dev/shm devpts none 0% 109.21 MB 0.00 KB 109.21 MB
/boot ext3 /dev/hda1 8% 85.33 MB 8.29 MB 98.72 MB
/dev/shm tmpfs none 0% 109.21 MB 0.00 KB 109.21 MB
/dev/shm tmpfs none 0% 109.21 MB 0.00 KB 109.21 MB
Totals : 4% 32.95 GB 1.28 GB 36.06 GB






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Created by phpSysInfo-2.5.2_rc1 on Jul 15, 2006 at 09:40 AM 0.6223 sec



************************************************** ********
Arsterisk info
************************************************** ********

Asterisk Info: asterisk1.local (192.168.0.6)

Version
Asterisk 1.2.6 built by root @ asterisk1.local on a i686 running Linux on 2006-04-08 16:11:22 UTC
Verbosity is at least 3




Uptime
System uptime: 15 hours, 23 minutes, 15 seconds
Verbosity is at least 3




Active Channel(s)
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
217.10.79.23 ******* 3352c2457a8 01816/00000 unkn No
217.10.79.23 ******* 6a2a82f6105 01815/00000 unkn No
80.176.233.167 (None) 00101/00000 unkn No
3 active SIP channels
Verbosity is at least 3




Sip Registry
Name/username Host Dyn Nat ACL Port Status
rsipgateout/******* 217.10.79.*** N 5060 Unmonitored
bsipgateout/******* 217.10.79.*** N 5060 Unmonitored
104/104 (Unspecified) D N 0 UNKNOWN
103/103 (Unspecified) D 0 Unmonitored
102/102 80.176.233.*** D N 1024 OK (106 ms)
101/101 192.168.0.5 D 5061 Unmonitored
100/100 192.168.0.5 D 5060 Unmonitored
7 sip peers [6 online , 1 offline]
Verbosity is at least 3




Sip Peers
Host Username Refresh State
sipgate.co.uk:5060 ******* 105 Request Sent
sipgate.co.uk:5060 ******* 105 Request Sent
Verbosity is at least 3




IAX2 Sip Registry
Host Username Perceived Refresh State
Verbosity is at least 3




IAX2 Peers
Name/Username Host Mask Port Status
0 iax2 peers [0 online, 0 offline, 0 unmonitored]
Verbosity is at least 3




Subscribe/Notify
-= Registered Asterisk Dial Plan Hints =-
104 : SIP/104 State:Unavailable Watchers 0
103 : SIP/103 State:Unavailable Watchers 0
102 : SIP/102 State:Idle Watchers 0
101 : SIP/101 State:Idle Watchers 0
100 : SIP/100 State:Idle Watchers 0
----------------
- 5 hints registered
Verbosity is at least 3




Zaptel driver info
Chan Extension Context Language MusicOnHold
pseudo from-pstn en
Verbosity is at least 3




Conference Info
No active MeetMe conferences.
Verbosity is at least 3




Voicemail users
Context Mbox User Zone NewMsg
default 101 David 1
default 102 Home 0
default 104 Simon 0
default 100 Brian 0
Verbosity is at least 3




NTP peers
remote refid st t when poll reach delay offset jitter
================================================== ============================
+cache.eenet.ee 88.201.227.*** 3 u 570 1024 377 60.987 -0.248 1.012
+ntp1.arttoday.c 204.34.198.*** 2 u 611 1024 377 156.873 5.407 0.243
*mozart.musicbox 192.5.41.*** 2 u 853 1024 177 20.817 -0.550 0.026
LOCAL(0) LOCAL(0) 10 l 64 64 377 0.000 0.000 0.001

[mberlant]

You are possibly experiencing those "unpredictable results" that come with trying to use a CODEC that doesn't exist. You should comment that out and try again. Your server CPU, etc., are overkill for such a small system, so that is likely not the source of your problem. What you don't mention is your internet bandwidth. How much does your ISP say you have? How much have you measured?

[zz000mm]

Which Codec does not exist?

My bandwidth should be ok as we have a 4Mbps. The router has QoS with ports used by Asterisk been given priority.

Any idea why my physical memory was very high??

Thanks for the help

Kind regards

Brian

[bbbeavis]

Asterisk "reserves" most of whats available. Mine does the same thing but without the sound issues you have. Is that 4Mbps upstream, too? Your upstream is often much lower and you may be pushing it if you are connecting with ulaw or alaw and have other internet traffic, too.

[zz000mm]

Hi bbbeavis,

Yes, upstream may be half a Mbps. I'll have to check.
Presumably although I've set up FreePBX, I have to fully register the Digium codec g729 before it will work. Is it worth me registering it? Do you think that this will sort out my problems.

Kind regards

brian.

[bbbeavis]

Sounds like you have the same bandwidth that I have. Unless you're doing something else to suck-up your bandwidth I would not bother using a heavily compressed codec. You also lose the ability to fax (if your voip provider also supports it). g711U (ulaw) would be my choice or g726 if you don't fax. g729 is great if your running a lot of concurrent calls and your stretching your bandwidth, though (which it doesn't appear that you are). I think following mberlant's advice to remove g729 is proper as it is not activated until licensed. Also, his suggest about checking the actual bandwidth of your connection might be a good idea. My friend had a residential cable setup with the same specs as you but it appeared they provisioned him a the basic commercial bandwidth which was 1/4 the speed.

[zz000mm]

bbbeavis,

Thanks for your help.

My phone adaptor is a Sipura SPA2100 which has two lines configured to two asterisk extensions. The sipura can handle the following codec's

G711u
G711a
G726-16
G726-24
G726-32
G726-40
G729a
G723

My sip.config file is

[general]
register => ***********
register => ***********
externip=************
localnet=192.168.0.0/255.255.255.0
nat=yes
canreinvite=no

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
qualify=no ;new
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm

#include sip_nat.conf



I've set up my extensions and trunks in my sip_additional.config file which looks like


[104]
username=104
type=friend
secret=*****
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=104@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Simon <104>

[bsipgateout]
username=*****
type=peer
secret=******
qualify=no
nat=yes
insecure=very
host=sipgate.co.uk
fromuser=******
fromdomain=sipgate.co.uk
dtmfmode=info
disallow=all
context=from-pstn
canreinvite=no
authuser=*******
allow=ulaw
allow=alaw
allow=gsm


If I remove the G729 and just have ulaw and alaw, do I just have the settings in my sip.config file only and remove the codec references from my sip_additional.config file....or should they be in both or the other way around?

I do fax....although it has not been working lately for some reason.....!

Kind regards

Brian.

[bbbeavis]

Have you forwarded UDP ports 5060, 10000-20000 on your router to your Trixbox which is supposed to have a static IP address?

[zz000mm]

Oh yes.......
We have ditched a 3com router for a Dlink that was easier to port forward then have gone for a nice new Linksys when the Dlink developed a fault.
Ports are forwarded and what a headache some routers can be................grrrrrr

[bbbeavis]

Do you have a static external IP or are you using dydns.org (or similar)? Probably wouldn't connect to begin with if misconfigured that way but it was a thought. I use to have a similar problem and it centered on the rtp streams. Also try Munin to see how your memory is being used. The majority of it should be reserved by Asterisk with a good portion for cache.

Correct me if I'm wrong, but I think freePBX will replace your deleted entries in under sip_additional.conf. You know, you may have some corruption going on now because of that; I would try deleting the trunk and reconfigure a new trunk without g729. I have had that work (kind of like doing the same thing over and over and expecting a different result).