Into.the.Void. » VoIP

SIP Express router, mysql and utf8

kargig — Sun, 17 Feb 2008 19:32:11 +0000

Description:
There’s a small problem when using SIP Express Router (net-misc/ser on Gentoo) with mysql support and your mysql server uses utf8 as a default character set (gentoo’s latest versions use utf8 by default).

The problem:
One of ser’s scripts (ser_mysql.sh) can’t handle utf8 tables.
# ser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': foobar creating database ser ... ERROR 1071 (42000) at line 100: Specified key was too long; max key length is 1000 bytes

Solution 1 (remove utf8):
Change the character set to latin1.
The specified error can easily be “fixed” by editing /usr/sbin/ser_mysql.sh. Inside that script you will find a line like this (line 38):
TABLE_TYPE="TYPE=MyISAM"

replace it with:

TABLE_TYPE="TYPE=MyISAM,DEFAULT CHARACTER SET latin1"

The result:
# ser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': foobar creating database ser ...

Solution 2 (reduce some column sizes):
The following solution makes the script run but I have not personally tested if ser STILL works ok after the changes. Feel free to experiment and comment back on this:
# sed -i 's|domain varchar(128|domain varchar(100|g' /usr/sbin/ser_mysql.sh # sed -i 's|contact varchar(255|contact varchar(128|g' /usr/sbin/ser_mysql.sh

intracom netroute, asterisk and sipdiscount

kargig — Thu, 09 Mar 2006 13:34:47 +0000

I own a netroute2 and I have an asterisk at home to serve me as a pbx. I use it primarily for testing, and I only have a working sipdiscount trunk on it so far. What I wanted to do was plug a normal phone on the netroute and set it up so that I can call land lines in Greece for free through asterisk, using a sip trunk with sipdiscount.

Asterisk configuration

add to your /etc/asterisk/sip.conf
[sipdiscount] type=peer host=sip1.sipdiscount.com dtmfmode=inband ;allow-g726 canreinvite=no fromdomain=stun.sipdiscount.com username=USERNAME fromuser=USERNAME secret=PASSWORD ;qualify=yes [5000] ; netroute type=friend regexten=5000 callerid="netroute2" <5000> host=dynamic ; This device needs to register secret=PASSWORD allow=ulaw allow=alaw

add to your /etc/asterisk/extensions.conf
exten => 5000,1,Dial(SIP/5000,30,rm) exten => _7.,1,Dial(SIP/${EXTEN:1}@sipdiscount,30,rm) exten => _7.,2,Congestion exten => _7.,3,Busy

Netroute Configuration

Netroute is more complicated than asterisk. Why ? because there is no documentation at all. So it’s a trial and error kind of thing. My current working solution IS very very crude and NOT thoroughly tested.

Edit your /etc/call_route.conf and comment out the line that says:
plugin load gr

Also, make sure your last lines of this file look like these:
# default routes route_pattern add default/@ gr filt_last rl_rg_sip0 # note: default route is required route_pattern add default/! default filt_last rl_rg_sip0

Now go to the web interface of netroute and in the voip section add asterisk’s sip domain/realm and IP settings and the settings you entered in the section [5000] of asterisk’s sip.conf. You might even want to lower the registration interval, if netroute and asterisk is in the same lan the extra traffic is insignificant. Then go to dialplan configuration and click on “+Add pattern”. In the Pattern field add 7X and don’t tick any of the other boxes. In the Prefix field just add 7.
When finished, do a
/etc/init.d/checkpoint /etc/init.d/rc-voip restart

Now pick up a handset plugged into the first netroute’s FXS port and dial a string such as: 700302101234567
7 in the beggining is used by asterisk to send the call to sipdiscount trunk and 0030 is used by sipdiscount to call Greece. So in order to call 2101234567 we need to dial 700302101234567 from the netroute.
If you just want to dial Greek land line though sipdiscount change
exten => _7.,1,Dial(SIP/${EXTEN:1}@sipdiscount,30,rm)
with
exten => _7.,1,Dial(SIP/0030${EXTEN:1}@sipdiscount,30,rm)

Then you would only need to call 72101234567 from netroute.

Check asterisk’s log file for errors. A successful call should look like this:
Executing Dial("SIP/5000-49ab", "SIP/00302101234567@sipdiscount|30|rm") in new stack

no progress…

kargig — Fri, 09 Dec 2005 10:31:26 +0000

Since gcc 3.4 was marked stable on gentoo, I emerged it and even did an emerge -e world. It took me some time because there were more than 600 packages to rebuild. Anyway, mplayer still denies to work with some videos using -vo xv even after the recompilation with the new gcc.
That’s the error message:

X11 error: BadAlloc (insufficient resources for operation)?,?% 1 0 32%

MPlayer interrupted by signal 6 in module: vo_check_events

I’ll keep trying…

Btw, I read some rumours that Cisco is going to release callmanager 5 under linux. I hope that becomes a reality because the current callmanager 4.1(3) runs only on windows 2000 and there are times that the whole system freezes for no good reason, or times when IE crashes. The whole configuration of the callmanager is done through a web browser…in order to add a single device you need 10 clicky-clickies…I hope this changes too.

ser/openser and Cisco ata 188

kargig — Thu, 17 Nov 2005 08:17:41 +0000

I’ve recently installed a ser (SIP express router) in one of my machines (well, in fact it’s more like openser) mostly for self tutoring. The config file was quite a pain to tune. Many many options, many many modules which all look quite usefull, and you can get easily distracted from what you should be doing. Luckilly there are some HOWTO’s (but not complete) around the net.
What I find about ser/openser VERY stupid and annoying, is that while it’s tools are mostly written in bash they have the default password that comes with the installation hardcoded inside. It would be much easier if the installation procedure asked you for a username/password instead of the defaults “ser/heslo”. Anyway…you can change them later quite easily…but it’s still annoying.
A tool you might need is sipsak. It’s like a wrapper of some common commands a SIP administrator might frequently need.

Installing Cisco ata 188 and making it work for ser/openser was trully easy. You have to download the latest sip image (ata_03_02_01_sip_050616_a.zip) for the ata adapter from cisco (you need a password for Cisco’s site) and upload it to your phone (instructions are inside the zip file’s readme file). Inside the zip file that contains the image and some readme’s there are a few other executables that can be used for debugging it. What’s interesting for us linux users? Cisco provides binaries for almost all those tools inside the zip! In fact I upgraded my ata from linux

Then go to http://ip.of.ata.given.by.dhcp/dev and you get a very nice menu with quite a lot of options, and some monitors (ethernet, RTP stats) that did not exist in previous versions. Just fill in your username/password, display name and sip proxy server..and you are good to go. The device registers itself without any problems and I was able to make calls to ata 188 from/to Linphone, Kphone and Xten-Lite(both windows and Linux).

Here’s my testing ata 188: