What is going on? Remote phones dont connect now

Discussion in 'General' started by reynolwi, Jul 7, 2008.

  1. reynolwi

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    Ok im starting to get frustrated. First FOP broke asterisk when i updated it to fix the flashing problems. So i reinstalled elastix and updated it to 1.1-7, but DID NOT update FOP and i went to start creating extensions and the remote phones wont connect anymore. I looked in the logs of the router and now its apperantly decided to start blocking it calling it a Port Scan Attack...

    07/07/2008 04:40:14.90 Port Scan attack !!! 10.25.18.103:5060 74.192.31.9:5060 UDP

    I have the following ports forwarded to elastix...

    Ports 5004-20000 to 10.25.18.13 (which is elastix) and its enabled. I even stuck elastix in the DMZ and that didnt work and then i tried a test phone and the stupid router still says Port Scan Attack.

    Is my router dying? Cause the remote extensions were working perfectly before i decided to update FOP and now they wont even connect. Im thinkin FOP and my router conspired against me.
     
  2. telecomtechnician

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    Re:What is going on? Remote phones dont connect no

    Hi there

    You should paste the following in this file sip_additional.conf

    Elastix 1.1.7 changed a lot of things, we have to be carefully once we do an update.


    bindport = 5060 ; Port to bind to (SIP is 5060)
    bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    disallow=all
    allow=ulaw
    allow=alaw
    externip=190.39.166.155
    localnet=192.168.0.31/255.255.255.0
    ; If you need to answer unauthenticated calls, you should change this
    ; next line to 'from-trunk', rather than 'from-sip-external'.
    ; You'll know this is happening if when you call in you get a message
    ; saying "The number you have dialed is not in service. Please check the
    ; number and try again."
    context = from-sip-external ; Send unknown SIP callers to this context
    callerid = Unknown
    tos=0x68
     
  3. telecomtechnician

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    Re:What is going on? Remote phones dont connect no

    externip=XXX.XXX.XXX.XXX
    localnet=XXX.XXX.XXX.XXX/255.255.255.0

    change this to the corresponding network parameters in your network

    Waiting for your comments

    David Medina
     
  4. reynolwi

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    Re:What is going on? Remote phones dont connect no

    why should this be copied into the sip_additional file. I have it in the sip.conf file like a lot of people have said to do.

    the phone is atleast communicating with the pbx now im just getting this message

    <--- Transmitting (NAT) to 74.192.44.50:64938 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 10.25.18.103:5060;branch=z9hG4bK04e72cea6bea4c9d;received=74.192.44.50
    From: "User" <sip:30101@txreyn-sip.point2this.com>;tag=b2692681f68ef584
    To: <sip:30101@txreyn-sip.point2this.com>;tag=as5544e54e
    Call-ID: dae306f118eef405@10.25.18.103
    CSeq: 10001 REGISTER
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c485abb"
    Content-Length: 0


    I have the phone set to connect to the use the external hostname for elastix and i get this message and i tried using the public IP for elastix and it did the same thing
     
  5. telecomtechnician

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    Re:What is going on? Remote phones dont connect no

    Hi

    I am making a correction of my previous post, in this file: sip_general_additional.conf after the last line you should put the two lines externip and localnet with their values that go with your network.

    As I said before, elastix 1.1-17 changed a lot of things in the config files and I think it has add new config files.

    You are working with an IPPBX bundled with a web interfase and some other applications, so the acknowledge and approach must be different. If you were working only with pure asterisk, of course this lines should go at sip.conf.

    I can tell you that my elastix production server is working fine and have a lot of external registrations.

    Waiting for your comments

    David Medina
     
  6. reynolwi

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    Re:What is going on? Remote phones dont connect no

    and where is this sip_general_additional file located because i must be starring at it.
     
  7. reynolwi

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    Re:What is going on? Remote phones dont connect no

    ok i did what you said and put it in the sip_general_additional file and im still getting this message.

    <--- Transmitting (NAT) to 74.194.213.243:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 10.25.18.80:5060;branch=z9hG4bKfb0762d090e9e6c7;received=74.194.213.243
    From: "Will Reynolds (20101)" <sip:20101@txreyn-sip.point2this.com>;tag=3af425f33819b2de
    To: <sip:20101@txreyn-sip.point2this.com>;tag=as1020b7c6
    Call-ID: 4f4e200158c32bc3@10.25.18.80
    CSeq: 10001 REGISTER
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54458adc"
    Content-Length: 0


    It didnt help it at all. This is really making me want to just reinstall 1.0-7 and just forget new elastix upgrade because nothing is helping. Freepbx is running 2.4 and it was before the upgrade as well and the remote extensions were working then.
     
  8. reynolwi

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    Re:What is going on? Remote phones dont connect no

    i did the command sip show peer 20101 and got this...

    * Name : 20101
    Secret : <Set>
    MD5Secret : <Not set>
    Context : from-internal
    Subscr.Cont. : <Not set>
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : 20101@default
    VM Extension : *97
    LastMsgsSent : 32767/65535
    Call limit : 50
    Dynamic : Yes
    Callerid : "device" <20101>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Nat : Always
    ACL : No
    T38 pt UDPTL : No
    CanReinvite : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    LastMsg : 0
    ToHost :
    Addr->IP : (Unspecified) Port 0
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username:
    SIP Options : (none)
    Codecs : 0xc (ulaw|alaw)
    Codec Order : (ulaw:20,alaw:20)
    Auto-Framing: No
    Status : UNKNOWN
    Useragent :
    Reg. Contact :
     

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