Hi,
thanks to all
I solved the calls dropped problem, it was "resetinterval" parameter in
zapata.....now asterisk does not drop calls anymore.
I do not get the message:
WARNING[3503] chan_zap.c: Got restart ack on channel 0/6 span 1 with owner
anymore...but I get all the others.
I'm interested to understand why I many messages like:
WARNING[21314] chan_zap.c: Ring requested on channel 0/1 already in use
on span 1. Hanging up owner
How can a channel be already in use??? That means the channel is
busy...if it is so then it is all right...but maybe that shouldn't be a
warning but a notice or something else...should it?
TIA
Giorgio Incantalupo
Rich Adamson wrote:
Steve Davies wrote:
On 9/12/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
Steve Davies wrote:
> For the curious, can anyone tell me how this flag fixes the issue?
- I
> have seen the error before, but always assumed it was related to hung
> channels.
>
> Thanks,
> Steve
>
> On 9/12/06, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote:
>> Problema solved!
>>
>> Just put resetinterval=never inside zapata.conf
>>
>>
>> Giorgio Incantalupo
If memory serves correctly, I believe the parameter was added a couple
of years ago as a means / workaround for hung channels. At the time,
there was not any overwhelming evidence as why a channel would
occasionally hang. Some of the possibilities included unusual
interaction from the opposite end of the T1/E1, anomalies in the
dialplan, etc.
Now that a substantial amount of work / changes have been made relative
to PRI's and other internal asterisk code, there appears to be less
of a
need to reset.
A reasonable approach might be to apply the parameter and pay close
attention to channels that might be in some strange state. If none are
observed, then leave it.
Thanks for that. I have a customer who is using Asterisk 1.0.x, and I
am tempted to backport this fix from the 1.2.x code where it was
introduced.
From a personal perspective, I think I'd hold off on the back port and
devote that time towards testing the soon to be released version (now
in Trunk).
If you've watched the number and type of changes that have gone into
SVN Trunk in the last couple of months, it appears as though a
significant number of possible memory leaks, sip code, infrastructure
code, PRI code changes, etc, have been applied that would be
beneficial for all production systems. There also appears to be a fair
amount of work that will be needed to upgrade dialplan syntax (etc)
for the new release.
Best guess is that once the Trunk code gets past the beta testing
phase, it will likely be the asterisk code of choice for most/all
production systems.
Consider the above is only my $0.02 worth. ;)
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