[asterisk-users] SIP endpoints CANCEL when PRI receives Cause Code 31

[Richard Mudgett]

> We have an Asterisk server which connects to another Asterisk server
> acting as a PSTN gateway. This gateway machine has Digium TE210P card
> connected to a pair of PRIs.
> 
> For the most part, all is working well, however there are some
> specific
> telephone numbers that my users have attempted to call, but we unable
> to.
> 
> I set debugging on and determined that when the the gateway machine
> dials one of the numbers in question, we receive from the PSTN an
> ISDN
> cause code 31, which in my understanding is not an error. This is
> then
> passed back to the originating Asterisk server via IAX as progress.
>  It
> is then sent to the originating endpoint as a sip message 183
> 'Session
> Progress'.  2 seconds after this 183 progress message is sent, the
> endpoint sends a SIP CANCEL message and the channel is torn down.
> 
> I have the prematuremedia=yes and progressinband=never in the
> sip.conf
> file which looks like it could be a solution, however I believe that
> because we are getting ISDN Call Proceeding and a corresponding SIP
> 100
> Trying message that this setting has no effect.
> 
> I have tried from several different endpoint types with the same
> results. I have verified that the numbers in question are in fact
> operational.
> 
> Any suggestions?
> 
> Asterisk version is 1.8.7 on both hosts
> Dahdi version 2.5.0
> libpri version 1.4.12

You should upgrade Asterisk to at least v1.8.8.  A regression in the
Asterisk v1.8.7 ./configure script does not setup Asterisk to use
libpri correctly.  Most supplementary service features and a hangup fix
supported by that version of libpri do not get enabled.

Richard