SIP trunk problem

Discussion in 'General' started by leiw3248, Mar 26, 2010.

  1. leiw3248

    Joined:
    Apr 21, 2008
    Messages:
    476
    Likes Received:
    0
    Hello

    The following is information of SIP trunk, but it not working.

    Outgoing Settings:

    Trunk name: sip

    PEER DETAILS:
    host=1.2.3.4
    type=friend
    disallow=all
    allow=ulaw
    qualify=yes
    port=5060
    dtmfmode=rfc2833

    Register String:
    12345678:abcd@1.2.3.4/12345678

    It is connected:
    voip*CLI> sip show peers
    Name/username Host Dyn Nat ACL Port Status
    sip/12345678 1.2.3.4 N 5060 OK (5 ms)


    -----------------------------------------------------------------
    Outbound routes:

    Dial patterns: 8|.

    Trunk Sequence: sip/sip

    The attached file is the log in /var/log/asterisk/full

    Thanks ! http://forum.elastix.org/old_files/full_log.txt
     
  2. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    Did you notice in your log file:

    [Mar 26 13:38:57] WARNING[6192] chan_sip.c: Received response: "Forbidden" from '"IT" <sip:110@210.0.214.125>;tag=as36b22a4b'


    you need to check with your provider what is acceptable in your peer (outgoing) settings, obviously yours is incorrect and absolutely nobody apart from another Hutchison client in Hong Kong using exactly the same provider as you, can help you. (or maybe they just don't like Saint Francis :) :) )


    p.s.

    you do not need a registration for outbound calls, that is purely for inbound. do they work?
     
  3. leiw3248

    Joined:
    Apr 21, 2008
    Messages:
    476
    Likes Received:
    0
    Hello dicko

    My friend told me he can trunk with his company VoIP server, his company VoIP server is Zed-3.

    Thanks !
     
  4. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    Then I suggest you ask your friend how he did it.

    dicko
     
  5. leiw3248

    Joined:
    Apr 21, 2008
    Messages:
    476
    Likes Received:
    0
    I have access his VoIP server, but the setting not same as Elastix. His VoIP server trunk setting don't need enter so many things. So I don't know what I'm missing.

    Thanks !
     
  6. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    SIP "Session Initiation Protocol" is an internet protocol, if you know where the other end is, and he knows how to get to you, then to can try and make a connection.

    What happens next is the two parties connect and then agree on authority, this can be any number of ways including username/password but not limited to that protocol, maybe your provider needs another agreement.

    Apparently this is where you call is failing, so you and only you (or your friend) will be able to sort that out with your provider, there are technically dozens of ways to connect to a far end, not all allow you to make an ongoing phone-call.

    FreePBX and Elastix try to make that as simple as possible, it works for most folks but not all, if it doesn't work for , I'm sorry but the only guy who can help you is the guy who has successfully done what you attempt but in his case,successfully. If you can't find that guy, then I suggest you just have to become that guy :)

    dicko

    p.s.

    Maybe you will this will help :

    http://www.voip-info.org/wiki/view/SIP
     

Share This Page