[Asterisk-Users] Disposition Failed still happening

Aaron Daniel amdtech at shsu.edu
Thu Dec 15 13:53:25 MST 2005


Yeah, that's basically what's happening here too... our scenario:

incoming call from PSTN --ZAP--> asteriskA --SIP--> asteriskB --SIP--> Phone

asteriskA always shows answered for us, asteriskB sometimes shows 
answered, but it's usually failed... I've even gone through and started 
noop'ing the cause code to see what the server at least sees, and it 
always shows a code of 16...

Aaron


tracinet wrote:
> I should note that in the following scenario:
>
> incoming call from PSTN ---SIP---> asteriskA ---IAX2---> asteriskB 
> ---SIP---> SIP Phone
>
> The call log does show disposition ANSWERED on asteriskA, but FAILED 
> on asteriskB.
>
> On 12/15/05, *tracinet* <traci.asterisk at gmail.com 
> <mailto:traci.asterisk at gmail.com>> wrote:
>
>     I actually opened a bug report on this earlier this month:
>
>     http://bugs.digium.com/view.php?id=5918
>
>     I have tried a new SVN version from a few days ago and it still
>     showed as FAILED for me in the following scenario:
>
>     incoming call from PSTN ---SIP---> asterisk ---IAX2---> asterisk
>     ---SIP---> SIP Phone
>
>     At least now it appears that the billsec field is no longer
>     showing zero, but the FAILED disposition is annoying.
>
>     If I was a programmer I would happily jump in and see what could
>     be done.  Maybe in my free time I can squeeze in a lesson in C
>     sometime ;)
>
>
>
>     On 12/15/05, *Aaron Daniel* <amdtech at shsu.edu
>     <mailto:amdtech at shsu.edu>> wrote:
>
>         Is anyone else still having disposition failed showing up in
>         the cdr's
>         on 1.2.1?  I can't seem to figure out why asterisk would put
>         that in the
>         cdr's when the calls have in fact completed successfully 0.o
>
>         Aaron Daniel
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