asterisk / SIP calls dropping

Posted on 2014-07-13
Last Modified: 2014-11-12
Hi I have an issue where my calls in Asterisk will just drop randomly after a random amount of time,  asterisk seems to have a hissy fit and then the call just crashes out.  I have turn on advanced logging (Verbosity 5 Debug 5 SIP Debug on) and have attached a cut down version of the logs,  wondering if anyone could make any sense of them (Logs are just the tail of when twice this issue happened)
Question by:ncomper
    LVL 15

    Expert Comment

    Back up please --  Asterisk what, 1.4 or 11,  on what OS and through a Firewall ?  

    We need a few details before looking at log files --
    LVL 5

    Author Comment

    Hi sorry this is hosted on the Amazon AWS cloud using their Firewall using NAT.  I.e Default external IP address and an internal IP address. The Version is "Asterisk (Ver.".  Im a little lost seems that people find it tough to read these logs :(.   Its running on the default Freepbx instance of Ubuntu which I believe is 10.0.4
    LVL 15

    Accepted Solution

    I would start by talking with your provider and see what this series of messages is all about --  Way to many of them is the short log

    <--- SIP read from UDP: --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP;received=;branch=z9hG4bK76a29781;rport=5060
    From: <>;tag=as3604d0b8
    To: <>
    CSeq: 163 REGISTER
    Call-ID: 4785356a0d89474560bc2d29285711ff@
    WWW-Authenticate: Digest realm="", nonce="c8a5a79899c0695e85d5e1120f9bc995", algorithm=MD5, opaque="1759901906"
    Content-Length: 0
    LVL 5

    Author Closing Comment

    This did not 100% solve it but lead me down the right path

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