SIP Trunk for Iptel

角色

sip.conf

1. [general]
   
2. disallow=all
   
3. allow=ulaw,alaw,gsm,g729,iLBC,speex
   
4. nat=yes
   
5. canreinvite=no
   
6. insecure=port,invite
   
7. qualify=yes
   
8. dtmfmode=rfc2833
   
9. context=fron-desk
   
10. 
    
11. register => john:john_password@iptel.org/from-iptel-john
    
12. 
    
13. [iptel]
    
14. type=friend
    
15. username=john
    
16. fromuser=john
    
17. fromdomain=iptel.org
    
18. host=iptel.org
    
19. secret=john_password
    
20. context=from-iptel
    
21. 
    
22. [6001]
    
23. type=friend
    
24. secret=6001_password
    
25. host=dynamic
    
26. context=internal
    
27. 
    
28. [6002]
    
29. type=friend
    
30. secret=6002_password
    
31. host=dynamic
    
32. context=internal

複製代碼

extensions.conf

1. [iptel]
   
2. exten => _102.,1,Dial(SIP/iptel/${EXTEN:3},,)
   
3. 
   
4. [internal]
   
5. 
   
6. include => iptel
   
7. 
   
8. exten => 6001,1,Dial(SIP/6001,,r)
   
9. exten => 6002,1,Dial(SIP/6002,,r)
   
10. 
    
11. [from-iptel]
    
12. exten => from-iptel-john,1,Dial(SIP/6002,,)

複製代碼

Using the prefix 102 for sending outbound calls for iptel. For instance 1000 to check voicemail.

YH

角色

The short codes for making SIP URI calls should be okay in most cases with any problem via Iptel trunk. Sometimes Iptel will stop for a while and resume very quickly. If you want to increase the chances of mkaing SIP URI calls, why do not you use your own Asterisk server to make it?