[asterisk-users] PRI: sometimes Asterisk drop calls

Giorgio Incantalupo gincantalupo at fgasoftware.com
Tue Sep 12 08:36:34 MST 2006

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?


Giorgio Incantalupo

Rich Adamson wrote:
> Steve Davies wrote:
>> On 9/12/06, Rich Adamson <radamson at routers.com> 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 <gincantalupo at fgasoftware.com> 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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