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SIP Trunk for Iptel

本帖最後由 角色 於 2010-9-1 15:44 編輯

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. register => john:john_password@iptel.org/from-iptel-john

  11. [iptel]
  12. type=friend
  13. username=john
  14. fromuser=john
  15. fromdomain=iptel.org
  16. host=iptel.org
  17. secret=john_password
  18. context=from-iptel

  19. [6001]
  20. type=friend
  21. secret=6001_password
  22. host=dynamic
  23. context=internal

  24. [6002]
  25. type=friend
  26. secret=6002_password
  27. host=dynamic
  28. context=internal
複製代碼
extensions.conf
  1. [iptel]
  2. exten => _102.,1,Dial(SIP/iptel/${EXTEN:3},,)

  3. [internal]

  4. include => iptel

  5. exten => 6001,1,Dial(SIP/6001,,r)
  6. exten => 6002,1,Dial(SIP/6002,,r)

  7. [from-iptel]
  8. 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

Can you use sip url in asterisk to dial without the use of short code? Sometimes it works but most of the time it doesn't for iptel.

TOP

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?

TOP

SIP URL dialing is not so easy to configure.  Many people failed to do so.  Anyhow, there are many suggestion and solutions to make it work.  You may try the following.

http://mikepultz.com/2009/04/han ... ialing-in-asterisk/

I, myself have not tried before.  If you can make it, please let us know.

Thanks

TOP

Thanks. It is very good indeed. This is the reason we need to use APL instead of relying on GUI

TOP

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