Voice drops 1 second after call connection

Discussion in 'General' started by newboy, Feb 16, 2010.

  1. newboy

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    Hi

    I have recently installed Elastix 1.6, after i had hack attacks due to upgrading to FreePBX version 2.6. I did not realize the sip settings under it allows the anonymous calls to come through. I had Fail2ban installed as well. All my settings were changed by the hacker and i was not able to connect to the PBX at all.

    I formatted the system and installed the version 1.6, 32 bit. Installed all the updates to bring it up to FreePBX 2.6. Disabled the SIP settings in FreePBX.

    My issue is:

    I am unable to make calls between remote extensions. I am able to make calls between the local extensions. But when i call between the remote extensions the voice drops after 1 sec of connection and my internet connection also drops at the router end. I have to reboot the router to get the connection back up.

    I have opened the following ports
    TCP/UDP: 5060
    UDP: 10000 to 20000 verified in rtp.conf the port numbers are right

    Another thing i have noticed is that my extensions are not connecting through port 5060. Even the local extensions are not connecting through port 5060.

    Name/username Host Dyn Nat ACL Port Status
    XXXX/XXXX 123.456.789.123 D N A 32768 UNREACHABLE
    XXXX (Unspecified) D N A 0 UNKNOWN
    XXXX/XXXX 123.456.789.123 D N A 60850 UNREACHABLE
    XXXX/XXXX 192.168.0.9 D N A 31446 OK (115 ms)

    The two remote extensions come up as OK (XX ms) initially when i connect but after making a call between them they turn to UNREACHABLE.

    When i try to check the sip channels while the call is active, it shows that the call used ulaw.


    This is what my sip_general_custom.conf file looks like:

    language=au
    videosupport=yes
    allow=g729
    allow=g723
    allow=h261
    allow=h263
    allow=h263p

    defaultexpirey=600
    maxexpirey=3600
    rtptimeout=60
    rtpholdtimeout=120

    alwaysauthreject=yes

    useragent = Elastix


    This is how my sip_nat.conf file looks like:

    nat=yes
    externip=XXX.XXX.XXX.XXX
    localnet=10.1.0.0/255.255.0.0
    externrefresh=10


    Any suggestions on what could be wrong here?
     
  2. fasilkaks

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    hi newboy,
    what trunk are you using for remote connectivity - sip or iax?
     
  3. fasilkaks

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    one more thing...
    could you paste the trunk settings here? both remote and local
     
  4. newboy

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    I am using a sip trunk. It is the same extensions that i am using remotely and or locally.

    here is the configuration for 1 of them:

    This device uses sip technology.

    secret
    dtmfmode: rfc2833
    canreinvite: no
    context: from-internal
    host: dynamic
    type: friend
    nat : yes
    port : 5060
    qualify : yes
    callgroup
    pickupgroup
    disallow
    allow
    dial : SIP/XXXX
    accountcode:
    mailbox : XXXX@device
    deny : 0.0.0.0/0.0.0.0
    permit: 0.0.0.0/0.0.0.0
     
  5. fasilkaks

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    why dont you go for IAX trunks...its very easy to setup and stable. I am using it for many months without failure.
     
  6. newboy

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    The IP phones i have only support SIP protocols :( so i can't use IAX extensions
     
  7. fmvillares

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    did you have the pbx in external public ip? did you have sip alg firewall enabled controlling that ip? do you use iptables or hardware firewaall, did you test if fail2 ban configs are correct to not block asterisk connections?
    remember that sip and nat are close enemies...
     

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