Posts tagged asterisk

Auto-provisioning with Asterisk and ST2030 Technicolor/Thomson phones

An introduction to the ST2030

The ST2030 is one of the few SIP phones distributed by Thomson (now changed name to Technicolor). In fact there are only 2 models: the ST2020, and the ST2030, and also a new one, the TB30, which is the successor to the ST2030. The ST2030 is supposed to have an End-of-Life set to the end of this year (2010), but I read that its EOL has been extended to the end of 2012.
In my personal experience, I think the ST2030 has the best price/functionalities/quality ratio. It has features like:

  • PoE (Power over Ethernet).
  • Headphone plug with a button on the phone to pickup with the headphone (or if you have a compatible headphone, pickup directly with a button on the headphone).
  • XML based Directory support, that you can interface with a remote HTTP server.
  • 4 differents lines/profiles (but not at the same time).
  • BLF (Busy Lamp Fields) to monitor other’s phone status (if they are using their phone, and even possibility to intercept a call).
  • Full compatibilty with Asterisk (tested on Asterisk 1.6+ and 1.8+).
  • Auto-provisioning with support for TFTP but also for HTTP/HTTPS, which simplifies quite a lot the provisioning configuration.

In this document, we’ll see the auto-provisioning process through DHCP+HTTP.

(more…)

Asterisk: DADHI module not working when using Xen

If you want to use any Asterisk module that needs a timer, like MeetMe, you have to use a module named dahdi (previously named zaptel). DAHDI has one module for each Digium supported card (B410P), and a dummy module (named dahdi_dummy) if you don’t have a hardware card, like me.

The problem appears when you have your Asterisk in a Xen environment. Xen does not allow the use of the RTC, so when using Dahdi/meetme, you get the following in you logs:

res_timing_dahdi.c: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection.

So get the sources, and let’s patch it!

svn co http://svn.digium.com/svn/dahdi/linux-complete/trunk DAHDI

In dahdi_dummy.c, you’ll have to comment the two defines USE_RTC, as in a Xen, you can’t use it:

# diff -u dahdi_dummy.c.ori dahdi_dummy.c
--- dahdi_dummy.c.ori	2009-03-23 09:50:36.000000000 +0000
+++ dahdi_dummy.c	2009-03-23 08:55:38.000000000 +0000
@@ -59,11 +59,11 @@
 #if defined(CONFIG_HIGH_RES_TIMERS) && LINUX_VERSION_CODE >= VERSION_CODE(2,6,22)
 #define USE_HIGHRESTIMER
 #else
-#define USE_RTC
+//#define USE_RTC
 #endif
 #else
 #if 0
-#define USE_RTC
+//#define USE_RTC
 #endif
 #endif
 #endif

Then compile the module, as usual, with :

/etc/init.d/dahdi stop
make all
make install
make config

Verify that your module has been correctly installed:

ls -al ./2.6.24-19-xen/dahdi/dahdi_dummy.ko

Comment out all the defined modules in the /etc/dahdi/modules file.

/etc/init.d/dahdi start
# dmesg
1007539.576458] dahdi: Telephony Interface Registered on major 196
[1007539.576468] dahdi: Version: SVN-trunk-r6201M
[1007540.642839] dahdi: Registered tone zone 2 (France)

Asterisk cirpack problem with Free and freephonie.net

If you configured your Asterisk/FreeSWITCH server to talk to your freephonie.net (french Free ISP provider), you’ll see in the logs the following warning message:
[Feb 12 09:29:06] WARNING[9228]: chan_sip.c:6624 determine_firstline_parts: Bad request protocol Packet

This is not really a problem, but an annoyance, as it fills up your logs. This is a known problem for more than years, but has never been corrected (neither by Asterisk nor by Cirpack devs). The usual correction was to add the following line to your startup scripts (/etc/rc.local on Debian for example):
iptables -A INPUT -p udp -m udp --dport 5060 -m string --string "Cirpack KeepAlive Packet" -j DROP

But the syntax has changed in iptables, and you’ll get the error:
iptables v1.3.6: STRING match: You must specify `--algo'

so, just add one of the 2 available algorithms (bm and kmp):
iptables -A INPUT -p udp -m udp --dport 5060 -m string --string "Cirpack KeepAlive Packet" --algo bm -j DROP

How to dial a number using Asterisk and Python

2

I didn’t find any example in Python on how to Dial a number from an Asterisk server and link it to another channel. So here’s a quick code to do it. You just need to have a manager defined in your /etc/asterisk/manager.conf defined.

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import socket
 
HOST="192.168.1.116"
PORT=5038
 
p = """Action: login
Events: off
Username: %(username)s
Secret: %(password)s
 
Action: originate
Channel: SIP/%(local_user)s
WaitTime: 60
CallerId: 600
Exten: %(phone_to_dial)s
Context: default
Priority: 1
 
Action: Logoff
"""
def click_to_call(phone_to_dial, username, password, local_user):
    pattern = p % {
            'phone_to_dial': phone_to_dial,
            'username': username,
            'password': password,
            'local_user': local_user}
 
    s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
    s.connect((HOST, PORT))
 
    data = s.recv(1024)
    for l in pattern.split('\n'):
        print "Sending>", l
        s.send(l+'\r\n')
        if l == "":
            data = s.recv(1024)
            print data
    data = s.recv(1024)
    s.close()
 
if __name__ == '__main__':
    click_to_call(phone_to_dial='123456789',
                  username='manager_login', password='yourpass',
                  local_user='600')
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