hold timeout

Discussion in 'General' started by witekprytek, Dec 22, 2009.

  1. witekprytek

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    I have noticed strange thing in my elastix box (e1.6).
    When caller wait in the queue is usually disconnected after 2 minutes of waiting. This same thing happens when I take caller on hold.
    For queue I have unlimited wait time. I do know where to set hold time-out.

    Could somebody help?
     
  2. witekprytek

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    I had made some debug and noticed, that no rtp packet are being send during phone is on hold.
    Asterisk notice, that user has ended session and disconnect

    <------------->
    Got RTP packet from 10.10.10.16:26552 (type 126, seq 005967, ts 000000, len 000004)

    <--- SIP read from 10.10.10.16:63176 --->
    BYE sip:aaaaaaaaa@10.10.10.20 SIP/2.0
    Via: SIP/2.0/UDP 10.10.10.16:63176;branch=z9hG4bK-d8754z-df11c6428e272610-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:629@10.10.10.16:63176>
    To: "aaaaaaaaa"<sip:aaaaaaaaa@10.10.10.20>;tag=as1a3f3a23
    From: "sip-kp"<sip:629@10.10.10.20>;tag=7e45ae70
    Call-ID: M2YxOThkMTJhY2VjM2Y3NGY4M2FmMjQ1NTNmZmJjYmM.
    CSeq: 4 BYE
    Proxy-Authorization: Digest username="629",realm="asterisk",nonce="68a13a20",uri="sip:aaaaaaaa@10.10.10.20",response="a330b7be6bf7bc5cde79cf409080f19e",algorithm=MD5
    User-Agent: X-Lite release 1100l stamp 47546
    Reason: SIP;description="User Hung Up"
    Content-Length: 0
     
  3. dicko

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    you should probably send them MOH ( or a least an occasional announcement, comfort noise is not implemented in Asterisk 1.4) or it is more than possible the rtp timers will expire on an inline router,agent or server and hence disconnect that session.
     
  4. dicko

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    WTF was that batman!
     
  5. witekprytek

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    Dicko
    This "disconnection" appears on XLite and grandstream phones. (I have made some test with polycom and SJphone and they work without any problem -maybe other vendors and models are affected too)
    X-lite after 30 seconds "on-hold" changes rtp payload type to 126 (?) without and asterisk will drop connection after this. RFC 3550 says that party should ignore unknown rtp payload: "A receiver MUST ignore packets with payload types that it does not understand" ...Asterisk 1.4.26 does not ignore it but just says "bye" :)

    With grandstream phones situation looks a little different - its look like GXP phones ignore "Sip Notify" when they become placed on hold.
    I have report this issue to the grandstream support with session debug - maybe they will help.

    Dicko - what batman do you mean?
     
  6. dicko

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    Apparently probably an error on Xlite's behalf

    http://forums.counterpath.com/viewtopic.php?f=6&t=12552

    Asterisk does indeed ignore the 126 packets as they were never properly negotiated, but in the absence of "valid" keep-alive packets in the rtp stream, it will time out and drop the connection after rtptimeout of the invalid packets (you can change the various rtp timeouts, check http://voip-info.org).


    (never mind the batman stuff it was the result of a brain-fart)
     

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