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

[Dale Noll]

On 05/23/2012 02:59 PM, Richard Mudgett wrote:
>> 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

Richard,

Thanks.  That appears to have done the trick.

Very much appreciated.

Dale

-- 
"The truth speaks for itself. I'm just the messenger."
      Lyta Alexander - Babylon 5