Lior Gradstein's Blog » voip http://www.gradstein.info Thu, 13 May 2010 09:46:29 +0000 http://wordpress.org/?v=2.9.2 en hourly 1 Asterisk: DADHI module not working when using Xen http://www.gradstein.info/software/asterisk/asterisk-dadhi-module-not-working-when-using-xen/ http://www.gradstein.info/software/asterisk/asterisk-dadhi-module-not-working-when-using-xen/#comments Thu, 26 Mar 2009 11:30:54 +0000 Lior Gradstein http://www.gradstein.info/?p=224 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)

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]]> http://www.gradstein.info/software/asterisk/asterisk-dadhi-module-not-working-when-using-xen/feed/ 5 Asterisk cirpack problem with Free and freephonie.net http://www.gradstein.info/software/asterisk/asterisk-cirpack-problem-with-free-and-freephonienet/ http://www.gradstein.info/software/asterisk/asterisk-cirpack-problem-with-free-and-freephonienet/#comments Thu, 12 Feb 2009 09:34:51 +0000 Lior Gradstein http://www.gradstein.info/?p=179 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

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]]> http://www.gradstein.info/software/asterisk/asterisk-cirpack-problem-with-free-and-freephonienet/feed/ 0 How to dial a number using Asterisk and Python http://www.gradstein.info/python/how-to-dial-a-number-using-asterisk-and-python/ http://www.gradstein.info/python/how-to-dial-a-number-using-asterisk-and-python/#comments Wed, 17 Dec 2008 09:39:17 +0000 Lior Gradstein http://www.gradstein.info/?p=33 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.

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+‘rn’)
        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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