asterisk at nicox.org wrote:
> This is working now, but asterisk restart every 10 to 15 minutes with this
> error:
>
> srv*CLI> *** glibc detected *** double free or corruption (out):
> 0xb6b1e0f8 ***
>
> Disconnected from Asterisk server
> /usr/sbin/safe_asterisk: line 50: 11910 Aborted (core
> dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY}
> </dev/${TTY}
> Asterisk ended with exit status 134
> Asterisk exited on signal 6.
> Automatically restarting Asterisk.
>
>
> any idea for debugging?
Oh yeah, it looks like you should have a core. Look in /tmp for it.
Matthew Fredrickson
>
>
> thanks
>
>
> Nico
>
> On Mon, 19 May 2008, asterisk at nicox.org wrote:
>
>> Hi Matthew,
>>
>> Great!, now it works as we, and hopefully other needs.
>>
>> thanks
>>
>> nico
>>
>>
>> On Fri, 16 May 2008, Matthew Fredrickson wrote:
>>
>>> asterisk at nicox.org wrote:
>>>> Hello Matthew,
>>>>
>>>> The problem is not solved with this patch, i think its really needed to
>>>> make an alerting, not only a progress.
>>>>
>>>> Please change that to be more Standard-compliant.
>>> I have not seen anything with clear definition for what this should be.
>>> The only thing that is somewhat useful that I have seen is Q.699. I
>>> just updated it again so that we do what Q.699 recommends (if called
>>> party's status in ACM is set to subscriber free, send ALERTING). It is
>>> not a very simple situation though, this may need further tweaking.
>>>
>>> Please update libss7 to rev 169 and Asterisk 1.6.0 branch to rev 116798
>>> to test this latest fix.
>>>
>>> Matthew Fredrickson
>>>
>>>> Thanks a lot
>>>>
>>>> Nico
>>>>
>>>>
>>>>
>>>> On Sat, 10 May 2008, Matthew Fredrickson wrote:
>>>>
>>>>> asterisk at nicox.org wrote:
>>>>>> Hello to everyone,
>>>>>>
>>>>>> We are using libss7 now for a long time, and we have seen a small problem.
>>>>>> If we get a ACM message from the carrier, asterisk says that the call is
>>>>>> Proceeding now, and not Ringing as described in the documentation of the
>>>>>> SS7-stack from Intel(dialogic).
>>>>>>
>>>>>> I changed 2 lines in chan_zap.c to solve this.
>>>>>>
>>>>>> here the diff for asterisk SVN-Version 115577:
>>>>>>
>>>>>> 9646c9646,9647
>>>>>> < p->proceeding = 1;
>>>>>> ---
>>>>>>> p->alerting = 1;
>>>>>>> p->subs[SUB_REAL].needringing = 1;
>>>>>>
>>>>>>
>>>>>> I hope i did it in the right way, becuase i'm on it to learn c at this
>>>>>> time, really no PRO.
>>>>> I added a patch that I think should fix it in trunk rev 115598 or in
>>>>> 1.6.0 branch rev 115599. Can you update and confirm that my fix (it was
>>>>> a little different from how you did it) works for you?
>>>>>
>>>>> This has been a problem for a while, for a few other people, so if there
>>>>> is anyone else that has had any issues related to the audio channel not
>>>>> connecting after getting an ACM, please update if you can to confirm
>>>>> whether or not this issue has been resolved.
>>>>>
>>>>> Basically what I did was to queue a progress frame along with the
>>>>> proceeding frame (which we already queued) when we receive an ACM, which
>>>>> should open up the audio channel to anything you are bridged to.
>>>>>
>>>>> Thanks!
>>>>>
>>>>> --
>>>>> Matthew Fredrickson
>>>>> Software/Firmware Engineer
>>>>> Digium, Inc.
>>>>>
>>>>> _______________________________________________
>>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>>>
>>>>> asterisk-ss7 mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>>>
>>>> _______________________________________________
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>>>>
>>>> asterisk-ss7 mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>
>>> --
>>> Matthew Fredrickson
>>> Software/Firmware Engineer
>>> Digium, Inc.
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
>>> asterisk-ss7 mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-ss7 mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-ss7
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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