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? 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 >>>> >>> >>> _______________________________________________ >>> --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. >> >> _______________________________________________ >> --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 >
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