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 Thanks, Dale -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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