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[keith.hardee at gmail.com]

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

[fons.vanderbeek at 84-...]

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:

[keith.hardee at gmail.com]

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:

[rj2807 at gmail.com]

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: