Calling with softphone--> We don't hear each other

Discussion in 'General' started by kalderista, Oct 24, 2008.

  1. kalderista

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    Hello,

    I'm trying to stablish callings from a softphone (N-lite and BOL SIPPhone) remotelly via internet. I register OK with the romote elastix machine but after dial the number nothing hapens. The softphone seems to be waiting to stablish the calling.

    The extension is created by default parameters, and I repeate again, it's well registered.

    Any idea?

    Thanx
     
  2. jades

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  3. danardf

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    And into each remote extension, include nat=yes.
    If you have a router, you must make a NAT for the rtp port: 10000 - 20000.
    Choise a good codec for each extension. (aluw or gsm or speex...Etc).
     
  4. jgutierrez

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    yep that is correct, if you are still having that problem, try to use IAX2 extensions for remote users.

    Iax2 solves a lot of NAT issues :p

    remember that IAX2 will use porte 4569 instead of 5060 (SIP)
     
  5. mihpel

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    i currently have an elastix box behind a firewall which is behind a dsl router which means double nat from the Internet to the pbx.

    What i have done to make it work for clients connecting through the internet is forward ports 5060-5061(tcp/udp) and 10000-20000(udp) from the router to the firewall and also forward 5060-5061(tcp/udp) and 10000-20000(udp) from the firewall to the pbx.

    Also in sip_nat.conf i have :

    nat=yes
    srvlookup=yes
    externhost=my.domain.net "externhost=dns or externip=static ip"
    externrefresh=5
    localnet=10.0.0.0/255.0.0.0 " your local network "
    canreinvite=no
    Qualify=yes

    Of cource all the extentions have nat=yes and if the client is behind any nat have the router-firewall-whatever forward the above mentioned ports to the ip of the pc hosting the client software.

    Until Stun can be handled by asterisk the above should work just fine.
     
  6. danardf

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    I don't think that Asterisk used the "SIP TCP"! Maybe asterisk 1.6 :huh:
    So, you can put 5060 UDP only. ;)

    for exemple from my configuration, sip_nat.conf is different:
    Code:
    externip = 61.121.111.111  // @IP public only
    localnet = 193.107.20.0/255.255.255.0  // LAN network
    
     
  7. mihpel

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    ---- off topic -----
    danardf why are you using a routed subnet (aka not private) for your local net? is it just an example?
    ---- off topic -----

    part of RFC 3261 SIP: Session Initiation Protocol

    .....

    All SIP elements MUST implement UDP and TCP. SIP elements MAY
    implement other protocols.

    Making TCP mandatory for the UA is a substantial change from RFC
    2543. It has arisen out of the need to handle larger messages,
    which MUST use TCP, as discussed below. Thus, even if an element
    never sends large messages, it may receive one and needs to be
    able to handle them.
    ......
     
  8. danardf

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    No it's not an exemple!
    It's my real config, and it's work always from several years ago.

    From my router, there's only UPD for the SIP protocol and it's always good.
    And even if you are reason with your explain, i think that's not useful into the current configuration! (Asterisk 1.4 = SIP UDP only)
    I don't see why we use TCP if asterisk not use TCP!
    And more, XLite not use the SIP TCP....Only the commercial version.

    In all the cases,why not, yes you can put TCP/UDP. Does no change for the result. ;)
     

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