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
Asterisk version is 1.8.7 on both hosts
Dahdi version 2.5.0
libpri version 1.4.12