Crazy Boy
2006-May-23 22:52 UTC
[Asterisk-Users] Configure Voipjet.com content in Asterisk
Hi,
I am Chandramouli from India. I have successfully implemented Intercom
(Dialling within my office LAN) and Voicemail features using Asterisk. To
implement this, I am using X-Lite Softphone.
Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any
external hardware to implement pure VoIP solution.
Here I am sending my configuration file values:
Contents of IAX.CONF:
[102]
type=friend
username=102
secret=chandra
host=dynamic
context=tutorial
[109]
type=friend
username=109
secret=ravi
host=dynamic
context=tutorial
Contents of SIP.CONF:
[102]
type=friend
username=102
secret=chandra
host=dynamic
context=tutorial
[109]
type=friend
username=109
secret=ravi
host=dynamic
context=tutorial
Contents of EXTENSIONS.CONF:
[tutorial]
exten => 102,1,Dial(SIP/102,10)
exten => 102,2,Voicemail(u102)
exten => 102,3,Voicemail(b102)
exten => 102,4,Hangup
exten => 109,1,Dial(SIP/109,10)
exten => 109,2,Voicemail(u109)
exten => 109,3,Voicemail(b109)
exten => 109,4,Hangup
Contents of VOICEMAIL.CONF:
[default]
102 => chandra,Chandramouli,chandra@xyz.com,chandra@xyz.com
109 => ravi,RaviKanth,ravi@xyz.com,ravi@xyz.com
Configuration of X-Lite softphone:
Enabled: Yes
Display Name: Chandra
User Name: 102
Authorization User: 102
Password: chandra
Domain/Realm: 192.168.91.63
SIP Proxy: 192.168.91.63
Out Bound Proxy: 192.168.91.63
Use Out Bound Proxy: Default
Send Internal IP: Default
Register: Default
I am eager to use a service provided by "voipjet.com". But, before
going for registered version, I want to use their trail version. For that I have
registered with "Instant IAX Test Account" and got some values. My
user name is "crazymoonboy". The values are:
Chandramouli Panigrahi, here are your personalized Asterisk configuration
settings. Print them out and keep them in a safe place, but do not share them
with others. If testing or debugging with a softphone, FireFly is recommended
although it is only optimized for the ILBC codec. iaxComm also works.
VoipJet account number (username/UserID): 9333
Authorization code (password): a47769538c462223 (You should see an MD5 string,
if it is blank logout and login again)
Peer1 East Coast Server: 64.34.45.100
NAC East Coast Server II: 66.246.220.19
Mzima West Coast Server: 72.34.43.5
InterNap West Coast Server II (soon to be discontinued): 69.25.60.30
(Choose depending on your location)
(N.B. 216.118.117.46 has been discontinued due to problems)
For Asterisk@Home AMP see this screenshot. For the regular Asterisk PBX setup,
see below:
Asterisk PBX Step 1: Add the following lines to the end of iax.conf (found in
/etc/asterisk)
[voipjet]
type=peer
host= 64.34.45.100
secret= a47769538c462223
auth=md5
notransfer=yes
context=default
Step 2: Add the following to extensions.conf (found in /etc/asterisk)
; NANPA: North American Numbers dialed as 1 + area code
; For example, the New York Public Library is dialed as 12123400849
; 1 (North American call) 212 (New York area code) 3400849 (libary's phone
number)
; WORLD: International Numbers dialed as 011 + country code + number
; For example, the Tate Modern Museum in London, U.K. is dialed as
011442078878000
; 011 (International call) 44 (U.K. country code) 2078878000 (museum's
number)
; Finally, the number just before @voipjet in the Dial string is your VoipJet
userid #
exten => _1NXXNXXXXXX,1,SetCallerID(4153574000); Set your CallerID as a ten
digit number like this. See our FAQ
exten => _1NXXNXXXXXX,2,Dial,IAX2/9333@voipjet/${EXTEN} ; VoipJet.com NANPA
exten => _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit
number like this. See our FAQ.
exten => _011.,2,Dial,IAX2/9333@voipjet/${EXTEN} ; VoipJet.com WORLD
;Do not change IAX2/9333 in the above two lines!
Step 3A (recommended): Set your codec to G.711 ulaw for optimal sound clarity
and minimal transmission delay. In iax.conf (found in /etc/asterisk) locate the
codec section and include the following only.
disallow=all ; Prevent all codecs...
allow = ulaw ; ...except G.711 ulaw
Step 3B (recommended): Also in iax.conf, enable the jitter buffer. This section
is usually immediately below the codecs section.
jitterbuffer=yes ; Jitter buffer enabled...
dropcount=1 ; ...to drop at most 0.5% of VoIP packets
How should I proceed and modify my configuration files content according to
given values. Is there any extra hardware needed to implement this? How to
implement this? Please help me out.
Looking forward for your quick response. Thank you.
Regards,
Chandramouli
INDIA
---------------------------------
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+
countries) for 2?/min or less.
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Chris Blunt
2006-May-24 01:55 UTC
[Asterisk-Users] RE: Configure Voipjet.com content in Asterisk
Hi Chandramouli
Setting up VoipJet is quite simple really, you have done all the hard bit to
get you Asterisk config this far.
Firstly may I point out if you are posting your configuration to this list
you change your password information, as you have just given everyone access
to your account at voipjet.
Make the changes to your iax.conf as voipjet suggest, the config they give
you is generated for you and is not generic.
Then you will need to add some provision in your dialplan (extensions.conf)
to route your outbound calls.
Something like:
exten => _9.,1,SetCIDNum(123456432) ; This is your proper phone number
exten => _9.,2,Dial(IAX2/1082@voipjet/${EXTEN:1},45,tr) ;dials the number
What this does: To make a call dial 9 followed by the number and press dial
on x-lite. The first command sets your Caller ID number. The second line
strips the 9 from the beginning of your number and hands the call to voipjet
to terminate.
You will need to ensure that your users have access to the context in wich
you put these entries.
As voipjet are US based you will need to dial your numbers in a us format.
Ie. 312 xxx xxxx (for calling Chicago).
Hope this helps you out.
Chris