Asterisk 13.11.2 Unable To Register On CentOS 7 64bit

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Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled

[1006]

type=friend

username06

secret=mysecret

context=sip-phone

call-limit=1

callerid=”iuser” <1006>

disallow=all

host=dynamic

allow=all

any ideas?

Thanks,

Motty

4 thoughts on - Asterisk 13.11.2 Unable To Register On CentOS 7 64bit

  • Hi Motty,

    Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then “sip set debug on”.

    Now try to register again. At last, ” sip de debug off”.

    Examine tour console or full log file to find some clue ir send me back some trace.

    Cheers.

    El oct. 13, 2016 1:45 PM, “Motty Cruz” escribió:

  • Hello Victor,

    I did set debug on, but I don’t see any errors. I did tcpdump, client is trying to register: here is the header of a udp packet

    User Datagram Protocol, Src Port: 55300, Dst Port: 5060

    Session Initiation Protocol (REGISTER)

    Request-Line: REGISTER sip:pbx.mydomain.com:5060 SIP/2.0

    Method: REGISTER

    Request-URI: sip:pbx.mydomain.com:5060

    [Resent Packet: True]

    [Suspected resend of frame: 14]

    Message Header

    Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK7b2855394DB988BE

    Transport: UDP

    Sent-by Address: 192.168.1.37

    Sent-by port: 5060

    Branch: z9hG4bK7b2855394DB988BE

    From: “1006” ;tag=2859342B-CBC71460

    SIP Display info: “1006”

    SIP from address: sip:1006@pbx.mydomain.com

    SIP from tag: 2859342B-CBC71460

    To:

    SIP to address: sip:1006@pbx.mydomain.com

    SIP to address User Part: 1006

    SIP to address Host Part: pbx.mydomain.com

    CSeq: 1 REGISTER

    Call-ID: 6cbe37bb-cca69d70-85d0431d@192.168.1.37

    Contact: ;methods=”INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”

    User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.10.0689

    Accept-Language: en

    Max-Forwards: 70

    Expires: 90

    Content-Length: 0

    Sip.conf

    [1006]

    type=friend

    username=1006

    secret=mysecret

    context=sip-phone

    call-limit=5

    callerid=”iuser” <1006>

    disallow=all

    host=dynamic

    allow=all

    nat=yes

    Is NAT value set to yes OK? Servers is on public IP, client is on private network.

    Thanks, Motty

    From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Victor Villarreal Sent: Thursday, October 13, 2016 10:09 AM
    To: Asterisk Users Mailing List – Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.11.2 unable to register on CentOS 7 64bit

    Hi Motty,

    Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then “sip set debug on”.

    Now try to register again. At last, ” sip de debug off”.

    Examine tour console or full log file to find some clue ir send me back some trace.

    Cheers.

    El oct. 13, 2016 1:45 PM, “Motty Cruz” escribió:

    Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled

    [1006]

    type=friend

    username=1006

    secret=mysecret

    context=sip-phone

    call-limit=1

    callerid=”iuser” <1006>

    disallow=all

    host=dynamic

    allow=all

    any ideas?

    Thanks,

    Motty

  • Ok.

    Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of the Polycom hardphone. If this is true, then you have NAT issues.

    The REGISTER message are received by your PBX, but when respond, Asterisk send the next SIP message to the IP informed by the phone, that is the internal LAN address. The messages do not reach back to the hardphone.

    You need to setup a STUN server in the Polycom hardphone settings. Please, check the manual. Search in Google some public STUN server to put in the settings.

    Last, the idea behind the “sip set debug” command was view the complete SIP
    messages conversation, not search for an error.

    On NAT escenarios, remember:

    * The NATed phones need to know the public IP of the NATing router. Either by manual setting or by STUN protocol.

    * Reduce the time between REGISTERs attempt, if the client have a dynamic IP connection.

    * Use the “localnet” SIP settings in Asterisk, so the PBX can distingish what Network need contacted via NAT and what not.

    Cheers.