[Bristuff-users] "Cause 34" on free channel

[Gunnar Schaller]

Hi Steve,
I use a standard Bristuff package from Junghanns with no
modifications. Just downloaded the package and "./install". At the
moment I have version 0.3.0-PRE-1y-p, but earlier with 0.3.0-PRE-1y-k
it was the same. Linux kernel is 2.6.22-2-686.
Signalling is PTMP. Why do you think changing to PTP should help?
There is nothing but Asterisk on the ISDN circuit.
I also thried setting lbo=0 in zaptel.conf as suggested in
http://www.ip-phone-forum.de/showthread.php?t=140061
(German Asterisk discussion on the same topic as here) But didn't
help.
One thing is confusing me: After having the "cause 34" I did a "zap
show channel 1". Result is within other lines
PRI Flags: Call
Why "Call"? I think this has to be empty. Can anyone confirm?

Regards,
Gunnar



Wednesday, May 7, 2008, 10:31:45 AM, you wrote:

> Oh, which bristuff and which Zaptel version? I did not see that detail
> in any of your previous posts.

> Steve

> 2008/5/7 Steve Davies <davies147 at gmail.com>:
>> Hi, some thoughts/questions...
>>
>>  Are you compiling with MMX or SSE extensions enabled for Echo
>>  Cancelling in Zaptel? I remember having very strange behaviour when I
>>  tried this about a year ago. If you are, then perhaps try reverting to
>>  the default settings.
>>
>>  Do you change any settings at-all from the default Zaptel+Bristuff
>>  during the compile?
>>
>>  Can you change your ISDN signalling to PTP?
>>
>>  Is anything else connected to the ISDN circuit except the Asterisk box?
>>
>>  Cheers,
>>
>> Steve
>>
>>  2008/5/6 Gunnar Schaller <linux at nowin.de>:
>>
>>
>> > As requested again my problem. See also my previous postings:
>>  >
>>  > http://lists.three-dimensional.net/pipermail/bristuff-users/2008-February/000018.html
>>  >  http://lists.three-dimensional.net/pipermail/bristuff-users/2008-February/000019.html
>>  >  4-port HFC (Junghanns quadbri). Mostly everything works good and I can
>>  >  dial in and out. But sometimes I get "cause 34 - Circuit/channel
>>  >  congestion" on dialing out. Definitely there is a free B-channel...
>>  >  I see nothing with "pri debug", so there must be a Asterisk-internal
>>  >  variable telling Asterisk all B-channels are in use on this span.
>>  >  But I can dial in on this span from the outside. And AFTER dialing in
>>  >  it is also possible to dial out again. Another solution is to restart
>>  >  Asterisk.
>>  >  I'm willing to try everything to solve this problem. So please bomb me
>>  >  with hints :-)
>>  >
>>  >
>>  >
>>  >  Thanks,
>>  >  Gunnar Schaller
>>  >
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