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neteasyinc's picture
Joined: 2010/03/09
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I installed the thirdlane mte today to test out how well this will work for our hosted pbx clients. It seems there is something funny in the way my sip trunk needs to be configured, can anyone translate what this should be in the thirdlane configuration panel:

1. If you have a packaged version of FreePBX (trixbox, PBX-in-a-Flash, etc) it is highly recommended that you use the FreePBX module from the section above.
2. Print this page for reference before you start
3. Login to your server via the web interface using a browser
4. Click on Trunks > Add SIP Trunk
5. Outgoing CallerID: 0000000000 (10-digits only) The name you set here will NOT be sent when you call regular PSTN lines.
6. Maximum Channels: Enter the number of channels you have purchased, 4 by default
7. If you are closer to San Jose, CA, use "sjc" instead of "jfk" (New York, NY) in the settings below.
8. Dial Rules:
* 011|.
* 1732+NXXXXXX ;<-- Replace 732 with your area code
9. Outbound Dial Prefix: +
10. Trunk Name: VP-SIPJFKA
11. Peer Details:
* type=peer
* context=from-pstn
* host=jfk-primary.voicepulse.com
* qualify=5000
* allow=all
* canreinvite=no
* username=Your Login from the Credentials Page
* secret=Your Password from the Credentials Page
* dtmfmode=rfc2833
* rfc2833compensate=yes
* insecure=port,invite
* trustrpid=yes
12. User Context: leave blank
13. User Details: leave blank
14. Register String: Login:Password@jfk-primary.voicepulse.com (use the Login and Password from the Credentials page)
15. Click "Submit Changes"
16. Repeat steps 4-14, except use VP-SIPJFKB and jfk-backup.voicepulse.com in place of VP-SIPJFKA and jfk-primary.voicepulse.com. You now have redundant trunks to VoicePulse!
17. Click on Outbound Routes > Add Route
18. Route Name: VP-OUT
19. Dial Patterns: Insert pre-defined patterns for Toll-free, Long Distance, and International
20. Trunk Sequence: Select SIP/VP-SIPJFKA, Click Add, Select SIP/VP-SIPJFKB, Click Add
21. Click "Submit Changes". You now have an outbound route to VoicePulse which will try both trunks defined earlier for toll-free, long distance and international calls.
22. Click on Extensions > Add SIP Extension
23. Extension Number: 101
24. Display Name: John Doe
25. Outbound CID: "John Doe" <0000000000> (include the "" and <>)
26. Secret: The SIP password for the SIP phone that John Doe is using
27. Click "Submit". You how have an extension that users can reach by dialing 101. You should try to get your SIP device to register to your FreePBX server now using the extension number as the username and the secret as the password.
28. Click Inbound Routes > Add Incoming Route
29. DID Number: A phone number from your Numbers page (MUST use 11-digits: 17323395100)
30. Set Destination: Select the Extensions radio button and select John Doe <101>
31. Click "Submit". You have now created an inbound route that will send all incoming calls to your phone number to John Doe's phone.
32. Repeat steps 19-29 for each phone number to user mapping you would like to define.
33. Restart Asterisk
34. Test incoming and outgoing calls from John Doe's phone.