remote SIP extension .. NOT work

Discussion in 'General' started by sertody, Jan 30, 2011.

  1. sertody

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    Hello,
    I am a beginner using Elastix 2.0.3 - 64bit, can you help me to set up a remote SIP extension ?

    I followed the recommendations I found the forum, but I can't run my remote station Grandstream 2010,

    I set the PBX:

    sip_nat.conf
    nat = yes
    externhost = xxxpbx.dyndns-ip.com
    localnet = 192.168.1.0/255.255.255.0
    externrefresh = 10

    Routers which is 192.168.1.1 I opened the ports :

    SIP = 5004 / 5082
    rtp = 10000 / 20000

    I set the station gxp2010:

    sip server = xxxpbx.dyndns-ip.com
    sip user id = xxx
    authanticate id = xxx
    authanticate pw = xxx

    thanks in advance to anyone who wants to help me
     
  2. dicko

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    Welcome here sertody:

    you state:-
    .
    .
    Routers which is 192.168.1.1 I opened the ports :

    SIP = 5004 / 5082
    rtp = 10000 / 20000
    .
    .

    Did you not just open, but also forward them to the Elastix box?. Only UDP/5060 and UDP/10000:20000 are needed by default.

    Please read "Elastix Without Tears" a couple of time for background.

    dicko
     
  3. sertody

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    HI Dicko
    thank you for support.

    sorry I forget to insert the IP address of my box ;I mean :

    sip_nat.conf

    nat = yes
    externhost = xxxpbx.dyndns-ip.com
    localnet = 192.168.1.0/255.255.255.0
    externrefresh = 10

    Routers which is 192.168.1.1 I opened the ports :

    SIP = 5004 / 5082 to 192.168.1.111
    rtp = 10000 / 20000 to 192.168.1.111

    I set the station gxp2010:

    sip server = xxxpbx.dyndns-ip.com
    sip user id = xxx
    authanticate id = xxx
    authanticate pw = xxx

    thanks in advance to anyone who wants to help me
     
  4. dicko

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    If you have UDP/5060 and UDP/10000:20000 (notice the UDP, that is important, TCP is not used) so forwarded, and any SIP trunks you have are working, then I suggest you need to

    SIP set debug ip <the IP of your remote extension>

    at the Asterisk CLI to see what is going wrong, sorry if that is a bit of a handful, but there is no real way to otherwise diagnose the failure without guessing

    dicko
     
  5. sertody

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    Thanks Dicko

    yes I have downloaded the manual of Ben Sharif , it's nice and I knew a lot of my box which now works with 2 SIP trunks!
    .
    For now the problem is connecting a remote station GXP2010
    .

    I followed your information and this belowe is what I see in the CLI:
    .

    sip set debug ip xxxpbx.dyndns sip debug ip-ip.com

    SIP Debugging Enabled for IP: 127.0.0.1
    .

    what should I do now ?
     
  6. dicko

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    :) :)

    Watch the CLI output (or preferably the log file at /var/log/asterisk/full)
    sooner or later, your Asterisk box will try to talk to the extension or vice versa, it should be bidirectionl, if there is a conversation, you will see what is failing, check your authorities. If there is NO effective conversation then suspect your network, . . . both ends!.
     
  7. sertody

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    Sorry Dicko

    Sorry, I have written is NOT WORK but I wanted to say you do NOT REGISTER !
     
  8. collector

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    localnet = 192.168.1.0/255.255.255.255

    This should be the internal IP of your elastix box. 192.168.1.0 does not look right to me.
     

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