Patrick’s Blog(2)

FreeSWITCH 1.0.0 “Phoenix” has been released. Congratulations to Anthony, Mike, Brian and everybody who has contributed to make this happen! For quite some time I have witnessed FS in the making on irc and the mailinglist. The amount of effort that has gone into creating FS is quite astonishing. It’s all too easy to take great Open Source projects like FreeSWITCH for granted but let’s not forget about the time and money the team has sunk into this project. It would be great if the Community starts helping out (more) with documenting things on the Wiki, bugfixes, testing etc. I look forward to give FreeSWITCH another whirl and see what has changed since last time I messed with it.

A while back the excellent Dutch ISP Xs4all introduced a VoIP service for its subscribers. Here is the configuration to make Asterisk 1.2 work with Xs4all. But before we go into the config one remark. There is a problem with Asterisk and how the Cirpack switch that Xs4all uses sends DTMF tones. You have to patch the file rtp.c of the Asterisk source and rebuild Asterisk. If you use Fedora, RHEL or CentOS then you can use these (S)RPMs that already have this problem solved.

Search in rtp.c for the following lines:

[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},

and add this line:
[96] = {0, AST_RTP_DTMF},

Now let’s get on with the configuration.

In sip.conf add the following lines. Obviously you need to replace 08787xxxxx with your number and replace ******** with your password.

register => 08787xxxxx:********@sip.xs4all.nl/08787xxxxx

[xs4all-in]
type=friend
username=08787xxxxx
fromuser=08787xxxxx
fromdomain=sip.xs4all.nl
secret=********
host=sip.xs4all.nl
insecure=invite
context=from-xs4all
canreinvite=no
dtmfmode=inband
disallow=all
allow=alaw


In extensions.conf add something like the following (adjust to your needs):

[from-xs4all]
exten => 08787xxxxx,1,Dial(SIP/100,30,t)
exten => 08787xxxxx,n,Hangup()

I did not need to use “nat=yes” although my Asterisk box is behind nat. It may depend on the modem. This setup works with a Thomson SpeedTouch 716.

Tony Mountifield recently found out that the inband detection of DTMF in Asterisk 1.2 was broken when the tones were not perfect. Check the bug report here. Because Asterisk 1.2 is in security maintenance mode only, Tony’s fix will not be committed to the Asterisk 1.2 code tree. Either get the patch from the bugreport or if you are on an RPM based system get the upcoming respin of the RPMs at http://www.laimbock.com/asterisk/.

The FreePBX project has released version 2.3. This release fixes more than 250 bugs in 2.3 and earlier releases. It is also the first release to support Asterisk 1.4. Read more about FreePBX here and download FreePBX here

Microsoft’s Steve Ballmer of monkey dance fame showed an interesting little gadget at the 3GSM show in Barcelona. A Skype like VoIP application on a mobile phone running some Microsoft OS that uses WiFi to make free VoIP calls using the Internet. Read more at TheBusinessOnline.com.