Using the default traffic shaper works really well for simple qos needs. Set your defined rtp ports (e.g. udp 3000 and 3001) and udp 5060 and udp 4569 as high priority. Add any offending traffic (e.g. gotomeeting at udp 8200) as low or medium. Takes about 5 minutes and works like a charm!
This could be needed for any number of reasons, but I needed to do this to have two trixbox Pro servers live next to each other on the same LAN behind an IPCop firewall. They were reporting back the same IP address to the hybrid hosting source, so inbound connections to both were routing to the one server. Normally this is remedied manually by changing the externip in sip.conf, but that is set automatically with trixbox pro, and not an option.
You need to log into the firewall at the console or via ssh. You need to comment out one line to disable masquerading, and add a few more in its place:
nano /etc/rc.d/rc.firewall
#Individual machine on GREEN
/sbin/iptables -t nat -A POSTROUTING -s 10.0.1.99 -j SNAT --to-source 1.2.3.5
#all other machines on GREEN
/sbin/iptables -t nat -A POSTROUTING -s 10.0.1.0/24 -j SNAT --to-source 1.2.3.4
#all other machines on ORANGE
/sbin/iptables -t nat -A POSTROUTING -s 10.0.2.0/24 -j SNAT --to-source 1.2.3.4
#DISABLE MASQUERADE
# /sbin/iptables -t nat -A REDNAT -o $IFACE -j MASQUERADE
TFTP uses UDP packets. The client connects from a random port to port 69, then the server connects back to the original port. This does not survive NAT (network address translation), possibly not on either end. trixbox pro and many other phone systems use TFTP for provisioning phones, so this is a problem for remote phones. Here is a fix that allows tftp traffic through an iptables based firewall (IPCop in this example).
You will need to enable ssh on the firewall and connect a session. IPCop uses non-standard port 222, and can usually only be connected to from the internal (green) network.
nano /etc/rc.d/rc.network
add these two lines:
modprobe ip_conntrack_tftp
modprobe ip_nat_tftp
Save and exit.
For immediate effect, repeat the two lines at the command prompt, or reboot the firewall. I had to repeat this on both firewalls, as IPCop firewalls were on both ends.
If using trixbox pro, you also need to make a few minor changes to the configuration files for the phones in order for the phone to try the correct server. Change the sNNNN.trixbox.fonality.com settings to sNNNNx.trixbox.fonality.com where NNNN is your server number. I modified the file itself. Fonality recommends modifying the phone config on the phone once it is initially configured.
About my only problem with the Grandstream BT-101 IP phone (other than the dreadful speaker phone) is the odd sized port for the headset. After looking pretty hard for an affordable adapter solution I finally find one at the local Radio Shack. It works just fine for $3.99.
At SSG we’ve recently implemented their first outbound call center. It is based on Gnudialer, an asterisk based outbound dialing system. Gnudialer is a little tricky to set up initially, but works very well, and the developers are quite helpful via their irc channel. The trick is to read all the readme files, and read them again.
The system runs on flat mySQL tables, which makes it pretty easy to report on. It also includes a rudimentary crm that does the trick. You can customize the script the agents see, including updatable data from the crm.
The initial trial was successful, and they will be ramping up to 60 agents within the next few months.
Getting asterisk to start properly on Debian is pretty simple. I had a hard time with various methods described elsewhere, though.
In a terminal session:
sudo nano /etc/rc.local
# add the following BEFORE the last line (exit 0):
mkdir /var/run/asterisk
chown asterisk:asterisk /var/run/asterisk
/usr/sbin/amportal start
Exact configuration details: Ubuntu 6.06 (kernel 2.6.15-26-server) within VMWare server on Xeon hardware. Asterisk 1.2.9.1 Freepbx 2.1.3
I have seen complaints that Polycom SoundPoint IP phones do not work well with NAT. The problem is the port used for RTP gets closed on the firewall after a while and the phone doesn’t know about it. Then calls go through, but the user gets dead air. A relatively harmless workaround is to have the phone re-register every 30 seconds. Then the port refreshes at an interval low enough that the firewall never closes it.
I found a reference to a very simple fix and have had success with it: Simply edit the .cfg file for the phone (assuming the phone is getting its settings from TFTP or FTP) and change the following value: reg.1.server.1.expires=”" to reg.1.server.1.expires=”30″
If you have other configured lines, then you’ll also need to change reg.2.server.1.expires, reg.3.server.1.expires, etc.
I’ve also had problems with the RTP ports being used falling outside of the range specified on the firewall’s port mapping, which caused the same symptoms. The phone started at port 16844. I just just adjusted the firewall, but there are also settings on the phone to force the RTP port or port range used.
This worked for a Polycom 501, but should be the same for any of the SoundPoint IP models: 300, 301, 500, 501, 600, 601.
First of all, upgrading Trixbox is so easy to do it is amazing. To update from Trixbox 1.0 to 1.1.1 you go to an ssh terminal window and type the following:
rpm -del zaptel-modules-2.6.9-34.EL
rpm -del zaptel-modules-2.6.9-35.EL
yum -y update
trixbox-update.sh
trixbox-update.sh update
reboot
rebuild_zaptel
modprobe wcfxo [if you have zaptel hardware]
modprobe wcfxs [if you have zaptel hardware]
genzaptelconf
reboot
That being said, I have run into a few problems during upgrades.
Yesterday I tested a Linksys SPA-901 IP phone with Asterisk. It has a solidity to it that reminds me of old ma-bell type phones (It could be used as a weapon like in Casino). The exterior of the phone is pretty spartan, not even an LCD display at all. It is fully functioning, though, with a large bright message light that works perfectly with my asterisk system. Importantly: it sounds as good as any phone I’ve used – that means better than 99% of them. It uses the g.711u codec by default but does include many other codecs including g.729. It also supports call waiting with asterisk and has an intuitive status light that indicates a call on hold.
My only complaint is the lack of documentation for provisioning. If I were to deploy a bunch of these for a client, I would want to use a tftp or ftp server for configuration. The data sheet on this says that is an option, but I could not find any examples or instructions anywhere. Granted the internal web server configuration is a snap and the thing boots up very quickly, so it is not a show stopper, for now anyway.
To quickly get to the web page in configure it, make sure you have DHCP working on your network. Then plug in the ethernet cable, and power it up. After a a few seconds it will be booted up. The status indicator light should not be flashing. If it is double-flashing red, there is a problem with the network or cable (from experience with a bad cable).
Pick up the handset and type in ****, then 110. A robot will tell you the phone’s IP address. You can then put that address in your web browser and configure it. You can easily dial in the SIP settings, change the ring tone and volume, codec, etc. Surpringly you can view a call history with callerid in a nicer format than any little LCD display.
Overall I’d have to say I’m very impressed with it.
We got the new big iron server from Signate running today, and in testing found it still suffers when playing MOH to > 20 calls using MP3 files. Signate uses a database based configuration instead of the normal asterisk flat config files, so I had to change the procedure a bit to get native sound files working.
In SigMAN, Click on “PBX Configuration”, then on “Manual Editor”
Select “MusicOnHold.conf from the popup
Comment out what is in that file, and add:
[default]
mode = files
directory = /var/lib/asterisk/moh-default
random=yes
Hit the update button at the bottom of the window.
You need to clear out the old mp3 files and populate new directory
from a terminal:
rm /var/lib/asterisk/mohmp3/*.mp3
mkdir /var/lib/asterisk/moh-default
(copy transcoded files to that directory)
asterisk -r
restart now
moh show classes
moh show files
You should see the new files and the default class, and the music should be working.
This is really just a slight variation on what is covered here.
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