Dropped Calls After 15 Minutes

Discussion in 'Help' started by edisoninfo, Jun 27, 2010.

  1. edisoninfo

    edisoninfo Guru

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    Piaf
    Asterisk 1.6.x
    FreePBX 2.7.x
    Sip Trunks from cable company
    No TDM style cards
    Dell R210 Server
    2 NICS --- LAN 192.168.1.x
    \-WAN outside IP no router iptables filtered hard
    dual default gateways so vpn to router to lan works

    They have 10 sip trunks or "lines" from the cable company. Well, they have 10 phone numbers and 10 call sessions. It doesn't matter what number is dialed, they can just have 10 calls at a time. Only one number is published so all calls are coming in with that DID tagged to it. They rarely have over 4, one time they had 6 concurrent calls. So, low usage rates, so I don't think anything is getting overloaded.

    How do I go about tracking down the cause of dropped calls? Looking at the FreePBX call logs the average call is around 55 seconds long. I am getting reports from users that longer calls, say 15 minutes, are getting dropped. I will try to get more specific details tomorrow when they are open. I glanced at the /var/log/asterisk/full log files and see the ending of calls but I can't tell if they where hung up on purpose or dropped. Is there any way to tell this?
     
  2. Bitnetix

    Bitnetix Guru

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    You need to learn to use the debugging tools within Asterisk (such as increasing the logging level, turning on SIP debug, and so forth) for us to be of any help. Chances are you have a network firewall that's timing out a NAT connection while the call is in progress. That's my guess.
     
  3. edisoninfo

    edisoninfo Guru

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    As I mentioned, I will collect as much info from the logs and users this week as I can. I'm afraid this is going to be a tough one to catch since it doesn't happen all the time and I can not afford to sit there all week waiting for it to happen. Hopefully I can replicate it.

    I don't "think" it is a nat problem since the sip trunks do not use nat. The eth0 nic is connected live to the internet with iptables locked down to only the ip address of the cable company. I have turned off nat on each of the phones since they all exist on the same 192.168.1.x lan. Anything is possible tho.
     
  4. Bitnetix

    Bitnetix Guru

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    Not NAT then. Sorry - wasn't trying to imply that you didn't know what you were doing. :)

    SIP debugging is going to be the most likely way to figure out what's going on.
     
  5. edisoninfo

    edisoninfo Guru

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    It seems this 15 minute dropped call thing is due to something in the 1.6 version of Asterisk. I have found a little info on it and it is very repeatable. 15 minutes and wham! phone call dropped. I have not figured out how to fix it yet tho.
     
  6. Bitnetix

    Bitnetix Guru

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    Here's how you test it. Turn on as much debugging as you can, make a call that goes to music on hold and log 17 minutes worth of junk. It will be a big log file, but it might help.

    You might also want to ask the cable company what they're doing. They should be able to tell you which side tears down the connection (theirs or yours) which should help figure out if it's your problem or theirs.

    At this point, I'm leaning towards theirs. 15 minutes is a magic number in the telco world. I doubt it's a coincidence.
     
  7. edisoninfo

    edisoninfo Guru

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    I'll give that a try, but I'm starting to be convinced it is an Asterisk issue. I found several bug reports and such when Googling. One of them is https://reviewboard.asterisk.org/r/698/

    Not sure how to fix it yet, still researching, but I think I'm on to something.....

    It has to do with the 1.6.x branch and a new thing called session-timers. Appears to be a bug but not sure if it has been fixed yet or in what versions....
     
  8. Bitnetix

    Bitnetix Guru

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    I'm curious to see how it all turns out. Please make sure to update.
     
  9. JasonH

    JasonH New Member

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    I had this issue as well, and it turned out to be a SIP session timer issue. Since you're at that exact time limit, I would suspect the same thing. For me it was a setting in our IP phones that I had set it not to use timers. Then I added this line to the sip_custom.conf file "session-timers=refuse". Give those a try.
     
  10. edisoninfo

    edisoninfo Guru

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    Since starting this thread I have reviewed the phone logs and found calls ranging from 45 to 75 minutes. So, despite the client telling me they all die at 15, there are longer calls. I did set the session-timers=refuse tho just in case.