SIP trunk to provider question

Discussion in 'General' started by leiw3248, Mar 17, 2010.

  1. leiw3248

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    Hello

    We have username / password / hostname gave by provider and then followed Elastix without tear to set SIP trunk, however, it not work.

    The following shown from asterisk console:

    test_sip/1234 210.245.176.100 N 5060 Unmonitored

    What's Unmonitored ? and I see this sip trunk is green icon on FoP.

    Thanks !
     
  2. Megabyte

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    Hi, could you please paste to us the configurations you made it, of courses dont paste the values for username and password, anotherthing did you made a register like

    user:password@hostnameorip
     
  3. leiw3248

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    Hello

    Outgoing:
    trunk: test_sip

    username=6789
    type=peer
    secret=1234
    insecure=very
    host=210.245.176.100
    dtmfmode=rfc2833
    disallow=all
    allow=ulaw
    canredirect=no
    canreinvite=no

    Registration:
    6789:1234@210.245.176.100/6789

    --------------------------------------
    Outbound Routes

    Dial Patterns: 8|.

    Trunk Sequence: test_sip
     
  4. leiw3248

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    Anyone can help ?

    Thanks !
     
  5. leiw3248

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    Could you can help me for this trunking problem? please
     
  6. Patrick_elx

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  7. leiw3248

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    I don't want to see monitored, I just want to know why I can trunk with service provider.

    Thanks !
     
  8. dicko

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    You might not want to see it but you did in fact indicate that that was a problem, or at least a point of confusion, for you in your original post.

    you attempt to register with the server, I suggest you do as you were advised (qualify the outbound) and post the output of

    rasterisk -x 'sip show registry'

    and

    rasterisk -x 'sip show peers'


    If both show ok, there is something wrong with your setup that ONLY you and your provisioner can resolve.

    dicko
     
  9. leiw3248

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    [root@voip ~]# rasterisk -x 'sip show registry'
    Host Username Refresh State Reg.Time
    210.245.176.100:5060 1234 105 Registered Mon, 22 Mar 2010 09:09:02
    [root@voip ~]# rasterisk -x 'sip show peers'
    Name/username Host Dyn Nat ACL Port Status
    test_sip/1234 210.245.176.100 N 5060 Unmonitored
     
  10. dicko

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    and if you insert the

    qualify=yes as recommended?
     
  11. leiw3248

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    [root@voip ~]# rasterisk -x 'sip show registry'
    Host Username Refresh State Reg.Time
    210.245.176.100:5060 1234 105 Registered Mon, 22 Mar 2010 09:09:02
    [root@voip ~]# rasterisk -x 'sip show peers'
    Name/username Host Dyn Nat ACL Port Status
    test_sip/1234 210.245.176.100 N 5060 OK (6ms)

    ----------------------------------------------------

    Here is the sip trunk config:

    PEER DETAILS:
    host=210.245.176.100
    context=from-pstn
    secret=1234
    type=peer
    insecure=very
    username=6789
    disallow=all
    allow=alaw
    qualify=yes
     
  12. leiw3248

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    Still cannot dial out with this trunk when I press outbound routes 8 number with 8xxxx.

    Thanks !
     
  13. dicko

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    Then every thing is fine and dandy, registered for inbound have a qualified outbound who sweet can it get. The inbound calls work I assume, so all you have to do is fix your outbound routes, I suggest you read "Elastix Without Tears" at least three time more as it really doesn't seem to be taking with you :)

    perhaps it is time for you to start actually looking at what /var/log/asterisk/full is telling you

    dicko
     
  14. leiw3248

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    Hello

    How can I setup inbounds call for this trunk ?

    Thanks !
     
  15. Chilling_Silence

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    You might want to have a read of Elastix Without Tears, it details that and more :)
     
  16. dicko

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    That is an absolutely amazing question for someone who has posted more than 300 times for help here, and has I assumed read "Elastix Withou Tears " more than once.

    I am as they say "gob-smacked"
     
  17. leiw3248

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    I just tried input the phone number 6789 to inbounds DID and CID and set the destination to (A ip-phone). I used external mobile to call back 6789, the (A ip-phone not ring, but another ip-phone pick-up the call ...

    The Elastix has installed a Openvox A400P card with 3FXO, so the PSTN call through those 3 lines to destination 600 ring groups, this 600 ring groups that included above "
    another ip-phone pick-up the call".

    Thanks !
     
  18. leiw3248

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    It work, I just enter the 6789 in DID !
     
  19. dicko

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    I suggest everything you thought you knew is wrong, throw it all away, start at Page one of "Elastix Without Tears" and stop when the pages run out, do that four times, please post again AFTER you have done that.

    Elastix installed nothing, YOU did, that you still don't understand the from-zaptel context is your problem, sorry if I sound grumpy but you really, really need to RTFM (more than once)

    dicko

    [EDIT]

    Holy God an epiphany !!!! well done leiw3248 !!!
     
  20. leiw3248

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    Don't laugh me -_-

    I'm trying the context=from-trunk

    host=210.245.176.100
    context=from-trunk
    secret=1234
    type=peer
    insecure=very
    username=6789
    disallow=all
    allow=alaw
    qualify=yes
     

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