[Asterisk-Users] Configure Voipjet.com content in Asterisk

[Crazy Boy]

   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 at xyz.com,chandra at xyz.com
 109 => ravi,RaviKanth,ravi at xyz.com,ravi at 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 at 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 at voipjet/${EXTEN} ; VoipJet.com NANPA 
 exten => _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ. 
 exten => _011.,2,Dial,IAX2/9333 at voipjet/${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

 
		
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