[asterisk-users] sip show channels - gives a growing list of dead channels

[Raj Jain]

Asterisk SIP channels can hang for a variety of reasons such as
network errors, signaling malfunction and software bugs. These are
difficult to track down and sometimes the root cause is not even in
your control. In order to provide a sort of "garbage collection"
mechanism for such hung SIP channels, Asterisk 1.6 supports a
mechanism called as SIP Session Timers. You may want to give this
feature a shot. The instructions for configuring it are in sip.conf.

--
Raj


On Mon, Mar 10, 2008 at 5:13 PM, Keith Hardee <keith.hardee at gmail.com> wrote:
> I feel like I've seen that error before, but I did some quick testing
>  and was not able to produce the error.  CLI level was greater than 206
>  (many v's)
>
>  call        from            to                   hangup
>  Test 1    polycom      spectralink     polycom
>  Test 2    polycom      spectralink     spectralink
>  Test 3    spectralink  polycom         polycom
>  Test 4    spectralink  polycom         spectralink
>  Test 5   spectralink   spectralink     spectralink
>
>  I only did one test of each above because I am not in office (had
>  someone doing tests while I watched CLI).  I can test more when I get
>  back Thursday.
>
>  Thanks for input.
>
>
>
>
>  On Sat, Mar 8, 2008 at 2:59 AM, Fons van der Beek
>  <fons.vanderbeek at 84-it.com> wrote:
>  > Same problem over here................
>  >
>  >  I use KIRK-Telecom ip600v3
>  >  This only happens on calls between SIP en MiSDN, anyone any clue?
>  >
>  >  As far as i can see these dead calls  once in while occur  when the
>  >  remote party first hangs up (remote=MiSDN channel)
>  >
>  >  Keith do you also have error messages in the CLI when you open asterisk
>  >  by using asterisk
>  >  -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv ? (a lot of v)
>  >
>  >   -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
>  >
>  >  10.0.0.71 represents the IP number of internal phone
>  >
>  >  Keith Hardee schreef:
>  >
>  >
>  > > I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18
>  >  > Spectralink wireless IP phones.
>  >  >
>  >  > Most of the Spectralink phones have entries in 'sip show channels'
>  >  > that do not go away.  None of the other phones do this.
>  >  >
>  >  > Is there anyway to remove these entries without restarting Asterisk?
>  >  >
>  >  > Any ideas on what could be done to prevent this?
>  >  >
>  >  > Example output:
>  >  > xxx.xxx.xxx.xxx   541         14dd18886d1  00103/00102  0x0 (nothing)
>  >  >   No       Rx: BYE
>  >  > xxx.xxx.xxx.xxx   546         1e7c2fd84ab  00103/00102  0x0 (nothing)
>  >  >   No  (d)  Rx: BYE
>  >  > xxx.xxx.xxx.xxx   546         80f99ee6-6c  00103/00104  0x0 (nothing)
>  >  >   No       Rx: BYE
>  >  > xxx.xxx.xxx.xxx   546         0d9b184254b  00104/00102  0x0 (nothing)
>  >  >   No       Rx: BYE
>  >  > xxx.xxx.xxx.xxx   546         7fa08c964a1  00104/00102  0x0 (nothing)
>  >  >   No       Rx: BYE
>  >  > xxx.xxx.xxx.xxx   542         7088c6a7-db  00102/00104  0x0 (nothing)
>  >  >   No       Rx: BYE
>  >  > xxx.xxx.xxx.xxx   541         424cc109052  00104/00102  0x0 (nothing)
>  >  >   No       Rx: BYE
>  >  > xxx.xxx.xxx.xxx   541         225fe5130e5  00104/00102  0x0 (nothing)
>  >  >   No       Rx: BYE
>  >  >
>  >  > Thanks,
>  >  > Keith
>  >  >
>  >  > _______________________________________________
>  >  > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>  >  >
>  >  > asterisk-users mailing list
>  >  > To UNSUBSCRIBE or update options visit:
>  >  >    http://lists.digium.com/mailman/listinfo/asterisk-users
>  >  >
>  >
>  >
>  >  _______________________________________________
>  >  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>  >
>  >  asterisk-users mailing list
>  >  To UNSUBSCRIBE or update options visit:
>  >    http://lists.digium.com/mailman/listinfo/asterisk-users
>  >
>
>  _______________________________________________
>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>  asterisk-users mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Raj Jain

mailto:rj2807 at gmail dot com
sip:rjain at iptel dot org