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Thread: Odd disconnect?

  1. #1
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    Default Odd disconnect?

    Made a call to a nearby town around 9:30PM EST. Almost exactly 30 minutes into the call, it was dropped and I heard fast busy. Redial went right through. Has anyone else had this happen?

  2. #2
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    Default Re: Odd disconnect?

    Quote Originally Posted by dswartz View Post
    Made a call to a nearby town around 9:30PM EST. Almost exactly 30 minutes into the call, it was dropped and I heard fast busy. Redial went right through. Has anyone else had this happen?
    by any chance you're using your own ATA, if you do see this thread
    especially my post
    BTW, I asked for a VOIPo approved ATA and it has been working OK

  3. #3
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    Default Re: Odd disconnect?

    Hmm, interesting. I could swear I've had calls longer than 30 minutes before. Also, NAT is not an issue for me (I re-read that thread), and if you scan back in it, you'll see I mentioned that my gateway runs asterisk, so NAT is not involved.

  4. #4
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    Default Re: Odd disconnect?

    Quote Originally Posted by dswartz View Post
    Hmm, interesting. I could swear I've had calls longer than 30 minutes before. Also, NAT is not an issue for me (I re-read that thread), and if you scan back in it, you'll see I mentioned that my gateway runs asterisk, so NAT is not involved.
    for me, NAT was not an issue, it seemed to be the fact that the SIP server did not see my ATA (SPA1001) as a non RFC 4028 compliant one; furthermore, the SIP server did not refresh the session timer (send the re-INVITE) as called for by the RFC

    you could verify this by sniffing the SIP convo

    BTW, is your ATA a linksys/sipura one?

    the grandstream HT286 and its variants are the only few ATAs out there that support RFC 4028

  5. #5
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    Default Re: Odd disconnect?

    Uh, I don't have an ATA (I'm running asterisk, remember?)

  6. #6
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    Default Re: Odd disconnect?

    Quote Originally Posted by dswartz View Post
    Uh, I don't have an ATA (I'm running asterisk, remember?)
    oops, sorry, I did not have enough cafein

    I still have my doubt that it is a NAT problem

    anyways, do sip trace to see if asterisk supports session timer and who's responsible for the refresh

    if you want, you can post the trace here (after some editing to protect the identity of the innocent)

  7. #7
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    Default Re: Odd disconnect?

    I doubt it too, since NAT isn't involved for me. I will be making a 30+ minute call tomorrow, so I will fire up a sniffer first.

    Update: I tried calling my cellphone from the voipo number and after 30:16, it disconnected. Looking at ethereal trace now...

    Update2: I don't see the update request either, and can confirm a BYE sent from the other end around 1800 seconds into the call. Bad news: asterisk 1.4 (what I am running) does NOT support RFC-4028. Possibly good news: asterisk 1.6 does (and that is much more stable now, as well as being (mostly) supported by freepbx 2.5. I will give a try to 1.6 and report back.
    Last edited by dswartz; 09-01-2008 at 05:15 PM.

  8. #8
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    Default Re: Odd disconnect?

    Got asterisk 1.6 working (mostly, still a few minor issues), and made a 1-hour call from my cellphone to the voipo line. Analysis of the wireshark trace doesn't show any Session-Expires (unless I just wasn't looking in the right place.) If it isn't right, I don't know why I can make a 1-hour call and couldn't before

    Update: maybe wireshark just wasn't decoding the packets right? Dunno. I did find a reference for 1.6 that said to set the sip options to 'session-timeout=originate', which will flag my end (the UAC) as the refresher and will reply if the UAS sends the refresh. All seems well, but I have to concur, it seems the openser implementation voipo is using is not complying with the RFC...
    Last edited by dswartz; 09-02-2008 at 06:56 AM. Reason: tweaked something

  9. #9
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    Default Re: Odd disconnect?

    Quote Originally Posted by dswartz View Post
    Got asterisk 1.6 working (mostly, still a few minor issues), and made a 1-hour call from my cellphone to the voipo line. Analysis of the wireshark trace doesn't show any Session-Expires (unless I just wasn't looking in the right place.) If it isn't right, I don't know why I can make a 1-hour call and couldn't before

    Update: maybe wireshark just wasn't decoding the packets right? Dunno. I did find a reference for 1.6 that said to set the sip options to 'session-timeout=originate', which will flag my end (the UAC) as the refresher and will reply if the UAS sends the refresh. All seems well, but I have to concur, it seems the openser implementation voipo is using is not complying with the RFC...
    it's good to hear that it's working ok for you
    IMHO, it's openser config/handling of non-RFC4028 compliant UACs
    Cheers,
    -v

  10. #10
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    Default Re: Odd disconnect?

    So have we determined that the PAP2's can't do this successfully?

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